[asterisk-users] iLBC installation problem
Hi, I would like to install iLBC codec. I have found a HOW TO ... here http://www.voip-info.org/wiki/view/iLBC. Unlucky when I compile with make I get the following errors. [code] Generating embedded module rules ... [CC] codec_ilbc.c - codec_ilbc.o codec_ilbc.c:40:30: error: ilbc/iLBC_encode.h: Nessun file o directory codec_ilbc.c:41:30: error: ilbc/iLBC_decode.h: Nessun file o directory codec_ilbc.c:56: error: expected specifier-qualifier-list before ‘iLBC_Enc_Inst_t’ codec_ilbc.c: In function ‘lintoilbc_new’: codec_ilbc.c:66: warning: implicit declaration of function ‘initEncode’ codec_ilbc.c:66: error: ‘struct ilbc_coder_pvt’ has no member named ‘enc’ codec_ilbc.c: In function ‘ilbctolin_new’: codec_ilbc.c:75: warning: implicit declaration of function ‘initDecode’ codec_ilbc.c:75: error: ‘struct ilbc_coder_pvt’ has no member named ‘dec’ codec_ilbc.c: In function ‘ilbctolin_framein’: codec_ilbc.c:113: warning: implicit declaration of function ‘iLBC_decode’ codec_ilbc.c:113: error: ‘struct ilbc_coder_pvt’ has no member named ‘dec’ codec_ilbc.c: In function ‘lintoilbc_framein’: codec_ilbc.c:131: error: ‘struct ilbc_coder_pvt’ has no member named ‘buf’ codec_ilbc.c: In function ‘lintoilbc_frameout’: codec_ilbc.c:152: error: ‘struct ilbc_coder_pvt’ has no member named ‘buf’ codec_ilbc.c:153: warning: implicit declaration of function ‘iLBC_encode’ codec_ilbc.c:153: error: ‘struct ilbc_coder_pvt’ has no member named ‘enc’ codec_ilbc.c:162: error: ‘struct ilbc_coder_pvt’ has no member named ‘buf’ codec_ilbc.c:162: error: ‘struct ilbc_coder_pvt’ has no member named ‘buf’ make[1]: *** [codec_ilbc.o] Errore 1 make: *** [codecs] Errore 2 [/code] I have tried with Asterisk 1.4.29 and 1.6.2.0. Where is the problem? Thank -- BOINC manager http://boincmanager.altervista.org -- _ -- 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 Management API
Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol standard. Peter. -- _ -- 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] Having problems with BLF
Hi,
I'm having a problem getting a snom 300 to work with BLF (extension
222). I've set it to watch extension 220 in the function key config
pages as per the wiki (BLF, sip:2...@server.com) but I can't get the
light to come on when 220 is ringing. The SIP trace page doesn't show
anything coming from my PBX when 220 is ringing or in use. Any help
much appreciated as this has been driving me mad for the last 2 days!
Is this an asterisk config prob (Asterisk 1.4.21.2)?
John
Console output:
-- Executing [...@default:1] SIPAddHeader(SIP/221-08ddaf00,
Alert-Info:http://nohost;info=alert-internal;x-line-id=0) in new
stack
-- Executing [...@default:2] Dial(SIP/221-08ddaf00,
SIP/220||tT) in new stack
-- Called 220
Extension Changed 220[blf] new state Ringing for Notify User 222 (queued)
-- SIP/220-08e0b528 is ringing
core show hints
-= Registered Asterisk Dial Plan Hints =-
2...@blf : SIP/223
State:IdleWatchers 0
2...@blf : SIP/222
State:IdleWatchers 0
2...@blf : SIP/221
State:IdleWatchers 0
2...@blf : SIP/220
State:IdleWatchers 1
- 4 hints registered
sip.conf
[general]
...
allowsubscribe=yes
subscribecontext=blf
notifyringing=yes
notifyhold=yes
call-limit=99
limitonpeers=yes
[220]
type=friend
username=220
secret=x
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833
[221]
type=friend
username=221
secret=xxx
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833
[222]
type=friend
username=222
secret=x
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833
mailbox=422
vmexten=702
fromdomain=sip3.x.co.uk
[223]
type=friend
username=223
secret=x
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833
extensions.conf
[default]
include = blf
exten =
_2XX,1,SIPAddHeader(Alert-Info:http://nohost\;info=alert-internal\;x-line-id=0)
exten = _2XX,n,DIAL(SIP/${EXTEN},,tT)
exten = _2XX,n,Hangup
[blf]
exten = 220,hint,SIP/220
exten = 221,hint,SIP/221
exten = 222,hint,SIP/222
exten = 223,hint,SIP/223
--
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[asterisk-users] Deadlock in Asterisk 1.4.29.1
Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:51] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7660fc0' [Mar 5 12:07:02] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7671d98' [Mar 5 12:07:07] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' [Mar 5 12:07:22] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76621d0' [Mar 5 12:10:55] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76a2130' [Mar 5 12:11:44] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7c04788' [Mar 5 12:12:52] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7675918' [Mar 5 12:15:11] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76772b0' [Mar 5 12:15:36] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' This happen along the day and resulting in a freeze of Asterisk. I mean that I need to kill -9 the process to be able to restart it. There's one thing different with Asterisk 1.2.35 : no deadlock AVOIDED in warning level while Asterisk is freezed. Thanks for your help. Regards, Adrien .L -- _ -- 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] FAX configuration for DAHDI lines
Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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] MGCP FXO endpoint
I have a fxo endpoint installed in a Cisco router. I would like in my dialplan to get an extension call a telephone number through that fxo endpoint. Since with zaptel channels it is done like: exten = 0999,1,Dial(DAHDI/2-1/111) -- being 111 the phone number I want to call. I thought that for mgcp it would be the same, and I did: exten = 5200,1,Dial(MGCP/aaln/S0/SU3/0...@armario11/111) aaln/S0/SU3/0 -- is an endpoint at ARMARIO11 The problem is that asterisk detects try to find host ARMARIO11/111 instead of calling number 111 in that FXO port. Here is the debug: -- Executing [5...@internal:1] Dial(MGCP/aaln/S0/SU2/0...@ignacio-1, MGCP/aaln/S0/SU3/0...@armario11/111) in new stack [Mar 5 13:36:50] NOTICE[4659]: chan_mgcp.c:1753 find_subchannel_and_lock: Gateway 'ARMARIO11/111' (and thus its endpoint 'aaln/S0/SU3/0') does not exist [Mar 5 13:36:50] WARNING[4659]: chan_mgcp.c:3541 mgcp_request: Unable to find MGCP endpoint 'aaln/S0/SU3/0...@armario11/111' [Mar 5 13:36:50] WARNING[4659]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'MGCP' (cause 20 - Unknown) Is there any way to achieve that? Thank you very much. Regards -- _ -- 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] FAX configuration for DAHDI lines
Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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] FAX configuration for DAHDI lines
Hi Petr, I would like this fax to be reached from a public number. I will replace the existing analog phone and replace the same with a fax. Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 * -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Gelencser Sent: Friday, March 05, 2010 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX configuration for DAHDI lines Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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 -- _ -- 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] Caller ID in Asterisk
As far as I know, you should set up the callerid in the chan_dahdi.conf
with the usecallerid=yes and the callerid=8001234001 options where you
are setting the each channels.
Regards,
Peter Gelencser
2010.03.05. 7:54 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta:
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten =
_988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn’t help.
I have 8 lines configured as below and need the callerID of the
individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
*From:* Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thursday, March 04, 2010 6:36 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private
Call and Out of Area in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)
exten = 8001234004,n,Set(CALLERID(name)=Line 4)
exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was
not going through
Venugopal
*From:* asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
*Sent:* Thu 3/4/2010 5:53 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I
guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)
exten = 8001234003,n,Set(CALLERID(name)=Line 5)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not
put them in ${}.
Jimmy
-Original Message-
*From:* venui...@motorola.com
*Sent:* Thu, 4 Mar 2010 19:50:03 +0800
*To:* asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com, asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
*From:* asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thu 3/4/2010 3:59 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP
server.When i call from the Asterisk machine to the SIP server i am
not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf
file is attached.
Cheers
venu
Re: [asterisk-users] Caller ID in Asterisk
Hi,
Well, if you replicate the line that set the callerid for every extension than you can set each one manually.
Jimmy
-Original Message-From: venui...@motorola.comSent: Fri, 5 Mar 2010 14:54:56 +0800To: venui...@motorola.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten = _988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn’t help.
I have 8 lines configured as below and need the callerID of the individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 *
From: Gopalakrishnaiyer Venugopal-Q16770 Sent: Thursday, March 04, 2010 6:36 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting "Private Call" and "Out of Area" in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]static=yeswriteprotect=noautofallthrough=noextenpatternmatchnew=noclearglobalvars=nopriorityjumping=yesuserscontext=default
[globals]CONSOLE=Console/dsp ; Console interface for demo;CONSOLE=DAHDI/1;CONSOLE=Phone/phone0IAXINFO=guest ; IAXtel username/password;IAXINFO=myuser:mypassTRUNK=DAHDI/G1TRUNKMSD=1
[Internal]include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)exten = 8001234004,n,Set(CALLERID(name)="Line 4")exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]exten = s,1,Answerexten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten = _988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was not going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy GodboutSent: Thu 3/4/2010 5:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)exten = 8001234003,n,Set(CALLERID(name)="Line 5")
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them in ${}.
Jimmy
-Original Message-From: venui...@motorola.comSent: Thu, 4 Mar 2010 19:50:03 +0800To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer Venugopal-Q16770Sent: Thu 3/4/2010 3:59 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.comSubject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is attached.
Cheers
venu
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Re: [asterisk-users] FAX configuration for DAHDI lines
Then it's simple, set up an exten in the extensions.conf like exten = 123456789,1,Dial(DAHDI/2,,rtT) exten = 123456789,n,Hangup() replace the 123456789 with the public phone number and the DAHDI/2 with the channel you are using. Regards, Peter 2010.03.05. 14:07 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Petr, I would like this fax to be reached from a public number. I will replace the existing analog phone and replace the same with a fax. Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 * -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Gelencser Sent: Friday, March 05, 2010 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX configuration for DAHDI lines Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations. Regards Venugopal G -- _ -- 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 -- _ -- 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] Asterisk Management API
At an Asterisk CLI use the command manager show commands. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 5, 2010, at 1:50 AM, Peter Childs wrote: Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol standard. Peter. -- _ -- 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 -- _ -- 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] 30 mins GSM file
Score another top-notch tip for Tilghman!!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, March 04, 2010 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 30 mins GSM file On Thursday 04 March 2010 16:51:54 David @ULC wrote: I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? One of the nicest things about gsm files is that having no file header, you can concatenate multiple files and get the same effect as having played the series of files. Within the standard set of files is silence/10, which is 10 seconds of silence. Concatenate 180 instances of that file, and the result will be 1800 seconds (30 minutes) of silence. for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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] Playback in h extension
Hi people, I'm trying to execute the PlayBack command in the h extension... but it is not played... is it possible to do that?Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu Blackberry® 8520 con Movistar por 0 €. ¿A qué esperas? http://serviciosmoviles.es.msn.com/messenger/blackberry.aspx-- _ -- 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] SIP / Echo Cancellation
On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote: Very informative post Vinícius ! 2010/3/5 Vinícius Fontes vinic...@canall.com.br mailto:vinic...@canall.com.br - Chandrakant Solanki solanki.chandrak...@gmail.com mailto:solanki.chandrak...@gmail.com escreveu: Hello I have successfully compiled OSLEC for echo cancellation for DAHDI channel. Is there any way to do echo cancellation for SIP Channel. Is any, please suggest me.?? Thanks in advance.. -- Regards, Chandrakant Solanki Short answer: Maybe. Depends on the SIP device you're using. Long answer: *takes a deep breath* First you gotta understand why echo occurs. Every single call you've ever made on your life has echo. You can hear yourself when you're speaking. If that was not the case, it would feel like talking on a push-to-talk system. So echo is a natural and even desirable phenomenom. What makes echo unconfortable is when the echo is *delayed* too much. There's a number of causes for this to happen. First and foremost, sometimes a part of the signal you're transmitting is reflected back to you. That usually happens on the analog part of the system (analog phones as a whole, the handset of an IP phone, the headset connected to your computer's sound card, etc). When we're talking about VoIP, the latencies involved are much higher than a completely TDM system. There's the encoding latency, easily understood as the time the device takes to convert the analog signal (your voice) in RTP packets, then there's the transmission latency, inherent to any network, and so on. All those latencies add up to each other, making the total latency go skyhigh and making you hear your own voice delayed by some milisseconds - the infamous echo. Asterisk cannot cancel echo when the call is entirely IP, from an IP phone to another, for example. There's simply no need for that. That's because it's the device's job to cancel the echo caused by its own TX reflections or analog/digital conversions. On the other hand, Asterisk can and will cancel echo if you have a hardware echo canceller or a software based one, like OSLEC -- which is by far the best software echo canceller I've ever seen. Finally, in order to solve your problem, you'll need to check a few things. If the call is entirely VoIP, from one end to other, then the IP phones, ATAs, gateways, softphones, whatever, are the sole responsibles on cancelling the echo. You'll need to turn on echo cancelling on this devices or tweak its parameters. Also, don't forget that latency makes echo much worse. If you control the entire network between the two phones, you MUST set up a QoS policy in order to minimize the latency as much as possible. I've solved many echo problems by just implementing end-to-end QoS on the network. Lastly (I swear I'm finishing this essay right here :), if that's not your case and you're having echo issues calling from a SIP phone to an external number, double check if OSLEC is indeed set as the echo canceller on /etc/dahdi/system.conf and enabled with echocancel=yes on your chan_dahdi.conf. You can always check if the echo canceller is active on a certain DAHDI channel by issuing the command dahdi show channel XX on Asterisk CLI, where XX of course is the said DAHDI channel. That covers the nature of the echo problem well, but it doesn't actually explain why echo cancellation over IP is almost always a failure. Echo cancellation is an adaptive process. It continually tunes a model of the system which is echoing. If that modelling is to have any chance of success, the system it is modelling must be stable and linear. The key stability issue with a VoIP channel is jitter buffering. Any jitter buffer in the path between you and the place causing the echo is likely to adjust the timing in a dynamic way. This means the echo timing will dynamically change. Every time it changes the echo canceller training is going to blow up. Not just go a little off tune, but really blow up. If your echo canceller isn't good at catching this kind of thing you might well get howling. You have little or no control over these jitter buffers. You might have control over your local link, but links further downstream are very rarely under your control. The other stability issue related to packet loss. When something is used to fill in for a lost packet it will not carry the normal echo signal. When the echo canceller removes the predicted echo a nasty noise will usually result - i.e. packet loss, however well concealed by clever PLC algorithms, will result in awful noises. The key linearity issue is lossy compressing codecs. The PSTN uses lossy compression -
Re: [asterisk-users] Playback in h extension
Not possible. H exten is called by a hangup. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, March 05, 2010 8:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Playback in h extension Hi people, I'm trying to execute the PlayBack command in the h extension... but it is not played... is it possible to do that? Thanks, Anahi _ Anahi Ludueña _ ¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es gratis! http://serviciosmoviles.es.msn.com/hotmail/movistar-particulares.aspx -- _ -- 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] Having problems with BLF
Hi! I'm having a problem getting a snom 300 to work with BLF (extension 222). I've set it to watch extension 220 in the function key config pages as per the wiki (BLF, sip:2...@server.com) but I can't get the light to come on when 220 is ringing. The SIP trace page doesn't show anything coming from my PBX when 220 is ringing or in use. First try with Extension instead of BLF. Which Wiki page are you referring to exactly? For example: http://www.voip-info.org/wiki-Asterisk+phone+snom http://wiki.snom.com/Interoperability/PBX/Asterisk http://wiki.snom.com/Features/Extension_Monitoring Then do a sip debug on your PBX to see if Asterisk is sending the device state information. If it is then you need to check your network setup (and make sure 222 is registered to the PBX as you might have instructed that phone to refuse SIP messages from anyone else). SIP SHOW SUBSCRIPTIONS might also reveal some more details. Also: - It is not advisable to name your sip peers with 22x = phone numbers. Those are devices that deserve device names. These usernames are far too easy to guess for a brute force attack, and they will put you into trouble when you re-arrange your diaplan. - Maybe except for [222] you most certainly do not need the username= statements. It does not do what you think it does. ;- Philipp -- _ -- 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] Follow-up to CALLERID(num) not working
I sent a question yesterday about having problems setting the caller ID. I turned on pri debug for both a good and bad call and I see this in the good call [2010-03-05 05:58:20.743] [6c 0c 21 80 30 30 30 30 30 30 30 30 30 30] [2010-03-05 05:58:20.744] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [2010-03-05 05:58:20.744]Presentation: Presentation permitted, user number not screened (0) '00' ] and this is the bad one [2010-03-05 06:19:27.099] [6c 0c 21 c3 30 30 30 30 30 30 30 30 30 30] [2010-03-05 06:19:27.099] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [2010-03-05 06:19:27.100]Presentation: Number not available (67) '00' ] Clearly this is why the caller ID is not being set. My question is who is figuring out if the forth value in the [ ] is 80 (where it works) or c3 (where it does not work)? The number was not actually 00. The first call was made from a SIP phone registered to the Asterisk box with the PRI line and the second one was via an IAX trunk to the system with the PRI line. Is there some setting in some conf file that allows or disallows this behavior? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Deadlock in Asterisk 1.4.29.1
If you want to open a bug report the proper place to do it is at http://issues.asterisk.org/ Compile with DEBUG_THREADS and DETECT_DEADLOCKS (see make menuselect compiler flags). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Mar 5, 2010 at 6:20 AM, Adrien Lemoine alemo...@legos.fr wrote: Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:51] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7660fc0' [Mar 5 12:07:02] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7671d98' [Mar 5 12:07:07] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' [Mar 5 12:07:22] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76621d0' [Mar 5 12:10:55] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76a2130' [Mar 5 12:11:44] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7c04788' [Mar 5 12:12:52] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7675918' [Mar 5 12:15:11] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76772b0' [Mar 5 12:15:36] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' This happen along the day and resulting in a freeze of Asterisk. I mean that I need to kill -9 the process to be able to restart it. There’s one thing different with Asterisk 1.2.35 : no deadlock AVOIDED in warning level while Asterisk is freezed. Thanks for your help. Regards, Adrien .L -- _ -- 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 -- _ -- 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] Having problems with BLF
Yes- followed all 3 wiki instructions. Thanks for naming tips! Does this log help at all? Looks like the PBX isn't sending the SIP messages- I notice the previous NOTIFY messages said (queued)- does this mean anything? John PBX*CLI sip show subscriptions Peer UserCall ID ExtensionLast state TypeMailbox 192.168.13.114 222 3c26707958d 2...@default Idle dialog-info+xml none 1 active SIP subscription My sip trace for 222: PBX*CLI set debug peer 222 SIP Debugging Enabled for IP: xx.69.xx.yy:2064 -- Executing [...@default:1] SIPAddHeader(SIP/221-09c99e60, Alert-Info:http://nohost;info=alert-internal;x-line-id=0) in new stack -- Executing [...@default:2] Dial(SIP/221-09c99e60, SIP/223||tT) in new stack -- Called 223 Extension Changed 223[default] new state Ringing for Notify User 222 (queued) == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager.d/README.conf': Found == Manager 'john' logged on from 127.0.0.1 == Manager 'john' logged off from 127.0.0.1 -- SIP/223-09ca07c0 is ringing -- Got SIP response 603 Decline back from xx.xx.xx.xx [THIS IS DIALLED EXTENSION 223 NOT ACCEPTING CALL] -- SIP/223-09ca07c0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [...@default:3] Hangup(SIP/221-09c99e60, ) in new stack == Spawn extension (default, 223, 3) exited non-zero on 'SIP/221-09c99e60' Extension Changed 223[default] new state Idle for Notify User 222 (queued) PBX*CLI sip set debug off On 5 March 2010 14:42, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I'm having a problem getting a snom 300 to work with BLF (extension 222). I've set it to watch extension 220 in the function key config pages as per the wiki (BLF, sip:2...@server.com) but I can't get the light to come on when 220 is ringing. The SIP trace page doesn't show anything coming from my PBX when 220 is ringing or in use. First try with Extension instead of BLF. Which Wiki page are you referring to exactly? For example: http://www.voip-info.org/wiki-Asterisk+phone+snom http://wiki.snom.com/Interoperability/PBX/Asterisk http://wiki.snom.com/Features/Extension_Monitoring Then do a sip debug on your PBX to see if Asterisk is sending the device state information. If it is then you need to check your network setup (and make sure 222 is registered to the PBX as you might have instructed that phone to refuse SIP messages from anyone else). SIP SHOW SUBSCRIPTIONS might also reveal some more details. Also: - It is not advisable to name your sip peers with 22x = phone numbers. Those are devices that deserve device names. These usernames are far too easy to guess for a brute force attack, and they will put you into trouble when you re-arrange your diaplan. - Maybe except for [222] you most certainly do not need the username= statements. It does not do what you think it does. ;- Philipp -- _ -- 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 -- _ -- 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 1.4 Followme Question
I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.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] SIP / Echo Cancellation
- Steve Underwood ste...@coppice.org escreveu: On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote: Very informative post Vinícius ! 2010/3/5 Vinícius Fontes vinic...@canall.com.br mailto:vinic...@canall.com.br - Chandrakant Solanki solanki.chandrak...@gmail.com mailto:solanki.chandrak...@gmail.com escreveu: Hello I have successfully compiled OSLEC for echo cancellation for DAHDI channel. Is there any way to do echo cancellation for SIP Channel. Is any, please suggest me.?? Thanks in advance.. -- Regards, Chandrakant Solanki Short answer: Maybe. Depends on the SIP device you're using. Long answer: *takes a deep breath* First you gotta understand why echo occurs. Every single call you've ever made on your life has echo. You can hear yourself when you're speaking. If that was not the case, it would feel like talking on a push-to-talk system. So echo is a natural and even desirable phenomenom. What makes echo unconfortable is when the echo is *delayed* too much. There's a number of causes for this to happen. First and foremost, sometimes a part of the signal you're transmitting is reflected back to you. That usually happens on the analog part of the system (analog phones as a whole, the handset of an IP phone, the headset connected to your computer's sound card, etc). When we're talking about VoIP, the latencies involved are much higher than a completely TDM system. There's the encoding latency, easily understood as the time the device takes to convert the analog signal (your voice) in RTP packets, then there's the transmission latency, inherent to any network, and so on. All those latencies add up to each other, making the total latency go skyhigh and making you hear your own voice delayed by some milisseconds - the infamous echo. Asterisk cannot cancel echo when the call is entirely IP, from an IP phone to another, for example. There's simply no need for that. That's because it's the device's job to cancel the echo caused by its own TX reflections or analog/digital conversions. On the other hand, Asterisk can and will cancel echo if you have a hardware echo canceller or a software based one, like OSLEC -- which is by far the best software echo canceller I've ever seen. Finally, in order to solve your problem, you'll need to check a few things. If the call is entirely VoIP, from one end to other, then the IP phones, ATAs, gateways, softphones, whatever, are the sole responsibles on cancelling the echo. You'll need to turn on echo cancelling on this devices or tweak its parameters. Also, don't forget that latency makes echo much worse. If you control the entire network between the two phones, you MUST set up a QoS policy in order to minimize the latency as much as possible. I've solved many echo problems by just implementing end-to-end QoS on the network. Lastly (I swear I'm finishing this essay right here :), if that's not your case and you're having echo issues calling from a SIP phone to an external number, double check if OSLEC is indeed set as the echo canceller on /etc/dahdi/system.conf and enabled with echocancel=yes on your chan_dahdi.conf. You can always check if the echo canceller is active on a certain DAHDI channel by issuing the command dahdi show channel XX on Asterisk CLI, where XX of course is the said DAHDI channel. That covers the nature of the echo problem well, but it doesn't actually explain why echo cancellation over IP is almost always a failure. Echo cancellation is an adaptive process. It continually tunes a model of the system which is echoing. If that modelling is to have any chance of success, the system it is modelling must be stable and linear. The key stability issue with a VoIP channel is jitter buffering. Any jitter buffer in the path between you and the place causing the echo is likely to adjust the timing in a dynamic way. This means the echo timing will dynamically change. Every time it changes the echo canceller training is going to blow up. Not just go a little off tune, but really blow up. If your echo canceller isn't good at catching this kind of thing you might well get howling. You have little or no control over these jitter buffers. You might have control over your local link, but links further downstream are very rarely under your control. The other stability issue related to packet loss. When something is used to fill in for a lost packet it will not carry the normal echo signal. When the echo canceller removes the predicted echo a nasty noise will usually result -
Re: [asterisk-users] 30 mins GSM file
On Thu, 4 Mar 2010, Steve Howes wrote: On 4 Mar 2010, at 23:11, Steve Edwards wrote: On Thu, 4 Mar 2010, Steve Edwards wrote: On Fri, 5 Mar 2010, David @ULC wrote: I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? Record yourself thinking of the solution for 1/2 of an hour. Use sox to concatenate 6.9 copies of John Cage's 4'33 Get permission first.. S Considering that was a 1950's era composition, perhaps the copyright has already expired? j -- _ -- 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] FollowMe / Asterisk 1.4 Question
Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call back into the extension's Asterisk VM box Thanks in advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.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] Deadlock in Asterisk 1.4.29.1
Hi Moises, Thanks for the URL. I hope to have a feedback before open an issue. If there isn’t I will do that. Regards, Adrien .L De : Moises Silva [mailto:moises.si...@gmail.com] Envoyé : vendredi 5 mars 2010 16:02 À : alemo...@legos.fr; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Deadlock in Asterisk 1.4.29.1 If you want to open a bug report the proper place to do it is at http://issues.asterisk.org/ Compile with DEBUG_THREADS and DETECT_DEADLOCKS (see make menuselect compiler flags). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Mar 5, 2010 at 6:20 AM, Adrien Lemoine alemo...@legos.fr wrote: Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:41] DEBUG[7130] channel.c: Avoiding deadlock for channel '0xb7c04788' [Mar 5 12:06:51] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7660fc0' [Mar 5 12:07:02] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7671d98' [Mar 5 12:07:07] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' [Mar 5 12:07:22] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76621d0' [Mar 5 12:10:55] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76a2130' [Mar 5 12:11:44] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7c04788' [Mar 5 12:12:52] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7675918' [Mar 5 12:15:11] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76772b0' [Mar 5 12:15:36] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb76acb08' This happen along the day and resulting in a freeze of Asterisk. I mean that I need to kill -9 the process to be able to restart it. There’s one thing different with Asterisk 1.2.35 : no deadlock AVOIDED in warning level while Asterisk is freezed. Thanks for your help. Regards, Adrien .L -- _ -- 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 -- _ -- 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] 30 mins GSM file
- Jeff LaCoursiere j...@jeff.net escreveu: On Thu, 4 Mar 2010, Steve Howes wrote: On 4 Mar 2010, at 23:11, Steve Edwards wrote: On Thu, 4 Mar 2010, Steve Edwards wrote: On Fri, 5 Mar 2010, David @ULC wrote: I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? Record yourself thinking of the solution for 1/2 of an hour. Use sox to concatenate 6.9 copies of John Cage's 4'33 Get permission first.. S Considering that was a 1950's era composition, perhaps the copyright has already expired? j Haha epic thread. Now seriously. I'm not sure why you want to make a 30-minute gsm-encoded silent audio file, but I'm pretty sure there's a better way to accomplish what you want than doing that. If you explain us what you're trying to do, maybe we can help even more. Anyway... Audacity can easily create a blank file of any lenght. Export it as WAV and convert it to GSM using Asterisk with the file convert CLI command. Make sure you create the file with 8000khz sampling rate, 16-bit resolution and mono, otherwise Asterisk won't be able to play it. Well, but there's no audio on the file either... *head asplodes* -- _ -- 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] Hardware requirements question.
I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little MM Technology, Inc. da...@mandm-tech.com 704.882.9432 x3 704.882.0405 FAX -- _ -- 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] AMI logs
Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu Blackberry® 8520 con Movistar por 0 €. ¿A qué esperas? http://serviciosmoviles.es.msn.com/messenger/blackberry.aspx-- _ -- 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] iLBC installation problem
Hi! I would like to install iLBC codec. I have found a HOW TO ... here http://www.voip-info.org/wiki/view/iLBC. Unlucky when I compile with make I get the following errors. After you ran the script to obtain the iLBC code, you need to go into asterisk/contrib/scripts/codecs/ilbc and copy everything to asterisk/codecs/ilbc. The compile asterisk again. Do correct/update the Wiki where you feel it is necessary. [code] Generating embedded module rules ... [CC] codec_ilbc.c - codec_ilbc.o codec_ilbc.c:40:30: error: ilbc/iLBC_encode.h: Nessun file o directory codec_ilbc.c:41:30: error: ilbc/iLBC_decode.h: Nessun file o directory Philipp -- _ -- 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] Denial of Service Attack
Hi, I currently have a dedicated server with a hosting provider for my voip and the provider is currently experiencing a DOS attack. I have been looking at purchasing a number of servers and creating my own VOIP setup with redundancy built in. However, how I can design the system to ensure services remain online in the event a DOS attack is launched? I use Polycom phones which can connect to two sip servers, so would I simply have to take down the affected SIP server so that all calls are routed through the backup server? Or is there a better way of doing things? Many thanks Dan -- _ -- 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] NeoSpeech Asterisk?
I am working on a project where a caller would call my PBX and get a menu of categories, sub-categories, and even more sub-categories. The reason for this is to determine what text-file a user wants to read out a large pile of them. I need some help deciding how to impliment this system. Firstly, I need a web-interface to make it easy for employees to write these text-files and save them into the system without having to know anything about servers and whatnot. Then, I need a way to convert these text-files into standard asterisk supported audio files or text-to-speech on the fly, which ever is easier. We would like fast forward and rewind and speed control but it's not top priority. Thanks, Byron -- _ -- 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] InterPBX communication using SIP
OK Guys i got fixed the phones i was using were registered in both servers which is not good, once i removed them it started working! 2010/3/4 khalid touati khalidtou...@gmail.com Hi Guys, i am using the following config in pbx1: register = pbx1:endop...@172.16.200.175 pbx1%3aendop...@172.16.200.175 [pbx2] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.175/255.255.255.128 in pbx2: register = pbx2:endop...@172.16.200.176 pbx2%3aendop...@172.16.200.176 [pbx1] type=friend host=dynamic trunk=yes sercret=password context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.176/255.255.255.128 and i get the following in pbx1: -- Executing [18...@default:1] Dial(SIP/8029-b7413678, SIP/pbx2/8021||TWw) in new stack -- Called pbx2/8021 [Mar 4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite: Received response: Forbidden from 'Khalid Touati sip:8...@172.16.200.176 sip%3a8...@172.16.200.176;tag=as1dcf5ff2' -- SIP/pbx2-09cf4468 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8029-b7413678' status is 'CONGESTION' though i am using the same config in IAX and it's working fine, also it's in the same context (so i believe it's a context issue). -- Abdullah -- Abdullah -- _ -- 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] 30 mins GSM file
Sorry if you guys find this silly, for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm * /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done* I need to enter above lines in my root prompt ? for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm * /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ;* * * * * * * On Fri, Mar 5, 2010 at 4:36 AM, David @ULC ucoms2...@gmail.com wrote: I believe we GSM of 8 bit for Asterisk ? On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote: Record a muted channel for 30 minutes like this: exten = s,1,Answer(1) exten = s,n,Progress() exten = s,n,record(silence_long.gsm|1800|s) exten = s,n,hangup Above option looks easy. What I have to dial from soft phone to get this ? On Fri, Mar 5, 2010 at 4:21 AM, David @ULC ucoms2...@gmail.com wrote: I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -- _ -- 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] Hardware requirements question.
- David Little da...@mandm-tech.com wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? whistle Zenons?!? Those must be brand new on the market... :-) In all seriousness, yes, I would think that hardware should handle the calls. BUT, how much will you be spending on power? My quick Googling shows thats a pretty beefy box. For what you could save in power, buy a shiny little Intel Atom based or similar low power system. You'll save on your monthly electrical costs plus, you'll have headroom to do other telephony tasks and not have to worry about your system load causing poor voice quality. My $0.02 USD. I accept cash only. :-) --Tim -- _ -- 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] Denial of Service Attack
On Friday 05 March 2010 10:17:27 Dan Journo wrote: I currently have a dedicated server with a hosting provider for my voip and the provider is currently experiencing a DOS attack. I have been looking at purchasing a number of servers and creating my own VOIP setup with redundancy built in. However, how I can design the system to ensure services remain online in the event a DOS attack is launched? I use Polycom phones which can connect to two sip servers, so would I simply have to take down the affected SIP server so that all calls are routed through the backup server? Or is there a better way of doing things? Best possible method would be to distribute your servers across many different networks, such that a DOS against all of your servers is effectively a DOS on the entire Internet. Then it becomes an issue for your upstream provider(s). Your upstream provider(s) would need to be involved in mitigating a DOS attack, anyway. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Hardware requirements question.
On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. Gordon -- _ -- 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] Hardware requirements question.
On Fri, 5 Mar 2010, Gordon Henderson wrote: On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. What's a MHz? This sounds like a really old box he just happens to have laying around... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] 30 mins GSM file
On Fri, 5 Mar 2010, David @ULC wrote: Sorry if you guys find this silly, for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm * /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done* I need to enter above lines in my root prompt ? Yes. Your system will run much better DISPLAYING the above command IN your root prompt. I know I've said it before (and you ignored it then), but you really should invest just a smidgen of time actually learning Linux. However, if you to CREATE a 30 minute silent GSM file using the above command, type in AT your root prompt: for i in `seq 1 180` (press ENTER here) do (press ENTER here) cat /var/lib/asterisk/sounds/silence/10.gsm \ (press ENTER here) /var/lib/asterisk/sounds/30-minutes-of-silence.gsm (press ENTER here) done (press ENTER here) For extra credit, figure out how the example below works and why it is (slightly) better: ( for i in `seq 1 180` do cat /var/lib/asterisk/sounds/silence/10.gsm done ) /var/lib/asterisk/sounds/30-minutes-of-silence.gsm -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] State of 64 bits applications in Asterisk
Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel -- _ -- 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] FollowMe / Asterisk 1.4 Question
On Fri, Mar 5, 2010 at 9:33 AM, Cory Andrews ipcbc...@gmail.com wrote:
Is there a way to strip the normal features out of FollowMe (call
acceptance, etc), and just set followme up to to blind transfer any call to
an extension's associated cell number if it is not answered on the extension
after 4 rings? Users want followme calls to wind up in their cellphone
voicemail and I'm having some issues with ring/answer timing and Asterisk
wants to pull the call back into the extension's Asterisk VM box
Thanks in advance!
Why not just set up their extension to try ring their desk phone for 20
seconds, then dial their cell phone for 40 seconds? Something like this:
exten = 100,1,Dial(SIP/100,20)
exten = 100,2,Dial(DAHDI/g1/${CELL_NUM},40)
exten = 100,3,Hangup()
--
Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] Regarding - P-Asserted identity
Hi All, We have two servers, one server (SIP asterisk server) sending calls to the second server(has PRI) which goes our through the PRI's (using TE 412p). When the pprivacy is enabled: P-Asserted-Identity Header, privacy id are sent in the header of SIP invite packet to the second server, how can we identify this privacy and block the callerid as the call goes to the second server which has the PRI cards (TDM circuit)? I tried setCallerPres(prob) but it prohibits all calls, is there any way of identifying the calls with the privacy ON coming from the first server and then block only those calls? Server details:asterisk: 1.4.26.2 dahdi: 2.2.0.2 libpri: 1.4.10.1 Thanks for your help. Regards Sandesh -- _ -- 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] Hardware requirements question.
On Fri, 5 Mar 2010, Steve Edwards wrote: On Fri, 5 Mar 2010, Gordon Henderson wrote: On Fri, 5 Mar 2010, David Little wrote: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? Since I'm happy doing that (or something similar) on a 1GHz processor with 256MB of RAM, I'd suggest that your box is somewhat over-specced It'll keep the room warm though. What's a MHz? This sounds like a really old box he just happens to have laying around... Doh! :) Looks like I missed that bit! Wow - 1GB of RAM in an old 550 MHz Xeon box. I've just given one of these away too - only had 256MB of RAM though! Actually, I reckon it'll work just fine though - I do all my testing on a very old 550MHz VIA system, and have production boxes on 500MHz Geode boxes, so make sure the distro is as lean as possible and off you go... Gordon -- _ -- 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] AMI logs
AMI does not create any log files itself. If actions cause actions that would otherwise get logged then those of course are logged. You need to receive the response from your action to see if there were problems. The AMI protocol calls for sending action packets and then receiving a response or responses back from that action. The response will sometimes have useful information as to why something does not work. I do not update the voicemail.conf file but here is an example of updating manager.conf: action:updateconfig reload:no srcfilename:base_manager.conf dstfilename:manager.conf Action-00:newcat Cat-00:newuser Action-01:append Cat-01:newuser Var-01:secret Value-01:nottelling Or agents.conf: action:updateconfig reload:chan_agent.so srcfilename:base_agents.conf dstfilename:agents.conf Action-00:append Cat-00:agents Var-00:agent Value-00:,,Newest one Or queues.conf: action:updateconfig reload:app_queue.so srcfilename:queues.conf.base dstfilename:queues.conf Action-00:newcat Cat-00:testing Action-01:append Cat-01:testing Var-01:eventwhencalled Value-01:vars Action-02:append Cat-02:testing Var-02:eventmemberstatus Value-02:yes -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 5, 2010, at 8:10 AM, Anahi Ludueña wrote: Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved? More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why... Thanks, Anahi Anahi Ludueña ¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es gratis! -- _ -- 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 -- _ -- 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] State of 64 bits applications in Asterisk
I've used FFA briefly but successfully on Asterisk 1.6.2 x64. Jordan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 05 March 2010 17:00 To: Asterisk-Users Subject: [asterisk-users] State of 64 bits applications in Asterisk Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel -- _ -- 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 -- _ -- 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] Observation about DAHDI, FAX and Echo cancellation
Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1 outgoing call. Now, with the help of dahdi show channel, if I check channel 2: echo cancellation is ON. Then i check channel 1: it is OFF. Shouldn't both be turned off? Regards, Håkon Nessjøen -- _ -- 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] Uverse, Asterisk and SIP
On Wed, Mar 3, 2010 at 1:23 PM, Fred Posner f...@teamforrest.com wrote: On Mar 3, 2010, at 1:03 PM, sean darcy wrote: Well at least my RG doesn't let you use DMZplus _unless_ you've chosen dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh into my router from the internet. Anybody else got this working? sean What are the issues? First, do you have a public IP or private IP from the DHCP server. If it's private, then it's not set up correctly. If it's public, make sure you've updated your sip.conf with the public ip as an external address. ---fred http://qxork.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 The issues are that sip doesn't work, even though this same set up worked with POTS dsl. IAX does (but gives lousy audio quality) so I don't believe all udp ports are blocked. ifconfig on my linux router box shows the public address. I can ssh into that box from the outside. This is a dynamic address, so I use Register to set the incoming ip address. As far as I can tell the Register never gets to another asterisk box I can inspect. I will try setting the home router address in the office asterisk box to see if that works and try a call from office to home, even though it's not a long term fix. sean -- _ -- 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] Observation about DAHDI, FAX and Echo cancellation
- Håkon Nessjøen haa...@avelia.no escreveu: Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1 outgoing call. Now, with the help of dahdi show channel, if I check channel 2: echo cancellation is ON. Then i check channel 1: it is OFF. Shouldn't both be turned off? Regards, Håkon Nessjøen If I recall it correctly, Asterisk needs to listen to the CED tone to determine if that call is indeed a fax or data transmission. Try to put an Answer() on your dialplan. -- _ -- 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] Uverse, Asterisk and SIP
On Mar 5, 2010, at 1:01 PM, sean darcy wrote: The issues are that sip doesn't work, What does doesn't work mean? In / Out? Both? Do you have a sip trace? even though this same set up worked with POTS dsl. IAX does (but gives lousy audio quality) so I don't believe all udp ports are blocked. ifconfig on my linux router box shows the public address. I can ssh into that box from the outside. This is a dynamic address, so I use Register to set the incoming ip address. As far as I can tell the Register never gets to another asterisk box I can inspect. does your sip.conf show your external ip? I will try setting the home router address in the office asterisk box to see if that works and try a call from office to home, even though it's not a long term fix. sean With att uverse I set the firewall on the att router off, my internal router/firewall to get the public ip via dhcp (it will give you a public ip and not a private one), and then set the sip.conf parameters. ---fred -- _ -- 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] Having problems with BLF
PBX*CLI sip show subscriptions Peer User Call ID Extension Last state Type Mailbox 192.168.13.114 222 3c26707958d ...@default Idle dialog-info+xml none 1 active SIP subscription The phone is behind natted router on a private IP, the PBX is on a public IP; could this private IP in the subscriptions be the problem? John -- _ -- 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] NeoSpeech Asterisk?
Byron, How about Cepstral TTS software for text to speech playout. I use it for short messages ( less than 5 mins). -Irfan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Byron J. Lee Sent: Friday, March 05, 2010 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] NeoSpeech Asterisk? I am working on a project where a caller would call my PBX and get a menu of categories, sub-categories, and even more sub-categories. The reason for this is to determine what text-file a user wants to read out a large pile of them. I need some help deciding how to impliment this system. Firstly, I need a web-interface to make it easy for employees to write these text-files and save them into the system without having to know anything about servers and whatnot. Then, I need a way to convert these text-files into standard asterisk supported audio files or text-to-speech on the fly, which ever is easier. We would like fast forward and rewind and speed control but it's not top priority. Thanks, Byron -- _ -- 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 -- _ -- 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] app_confbridge production ready?
I have an existing conference bridge running on Asterisk 1.4.2 using MeetMe and it's been pretty much rock solid since it was installed. We do around 460,000 minutes on it monthly and peak at about 150 simultaneous sip channels. I'm adding a second bridge for redundancy purposes into another facility and would like to go with app_confbridge and Solaris. Since it's a relatively new app I have to question if it's been put to the test and whether I can expect the same kind of stability. Does anyone use confbridge in a large installation and can provide feedback on its stability, quality in comparison to MeetMe? I use a sangoma card in my 1.4.2 box to provide timing and it has never been an issue. Can I expect similar performance from the new timing API? Thanks Bob -- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- 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] Hardware requirements question.
Yes, this machine will be enough for that task. Performance wise. The other good thing is that it is not very likely that someone will steal your PBX. As far as I remember it is a 7 rack unit box which weights approx. one metric ton. ;-) But remember - if anything dies in the box and you have to get spare parts quick you will pay more than you want to. Chris 2010/3/5 David Little da...@mandm-tech.com: I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little MM Technology, Inc. da...@mandm-tech.com 704.882.9432 x3 704.882.0405 FAX -- _ -- 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 -- _ -- 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] Playback in h extension
2010/3/5 Danny Nicholas da...@debsinc.com: Not possible. H exten is called by a hangup. Well - sometimes not both parties hang up at the same time. ;-) If you want to play something to the originating party after die Dial()ed party hangs up use the option g in the Dial command to get more commands executed after the called party hangs up. There you could check the system variable DIALSTATUS to check if the called party ANSWERed the call or was BUSY etc. I hope that helps a bit. I just wrote it from the back of my mind. Please check the documentation of the Dial command. If you are not in a Dial() situation Danny's comment applies. ;-) Chris -- _ -- 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] Asterisk 1.4 Followme Question
Isnt that the point of the FMFM – to allow the call to come back into the asterisk server and have your voicemail managed in one location? If not wanted, I guess remove the voicemail step from the FMFM config and just have it end on the forwarded cellphone. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Cory Andrews *Sent:* Friday, March 05, 2010 10:16 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk 1.4 Followme Question I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbc...@gmail.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] Having problems with BLF
Hi! PBX*CLI sip show subscriptions Peer User Call ID Extension Last state Type Mailbox 192.168.13.114 222 3c26707958d ...@default Idle dialog-info+xml none 1 active SIP subscription The phone is behind natted router on a private IP, the PBX is on a public IP; could this private IP in the subscriptions be the problem? Not sure, haven't seen that before. Anyone? Do add a SIP SET DEBUG IP 192.168.13.114 to the game and see what is happening. What does the result of route -n look like? What does SIP SHOW PEER 223 say concerning Status and Addr-IP, can 223 be called? In general: You will want to accompany nat=yes with canreinvite=nonat (or canreinvite=yes). You could also turn on STUN in the snom and then switch to nat=no in Asterisk. Philipp -- _ -- 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] MOH Oddity
I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. When I throw two calls into a queue, they are both on different seconds of a song, or on different songs. Here is my MOH config: [general] cachertclasses=yes [Test] mode=files directory=/var/lib/asterisk/moh/Test/ [default] mode=files directory=/var/lib/asterisk/moh/ Any thoughts on what I might be doing wrong? -- _ -- 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] MOH Oddity
Forgot to include: I'm running 1.6.2.5 On Fri, Mar 5, 2010 at 5:29 PM, Matt mhop...@gmail.com wrote: I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. When I throw two calls into a queue, they are both on different seconds of a song, or on different songs. Here is my MOH config: [general] cachertclasses=yes [Test] mode=files directory=/var/lib/asterisk/moh/Test/ [default] mode=files directory=/var/lib/asterisk/moh/ Any thoughts on what I might be doing wrong? -- _ -- 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] Having problems with BLF
Clarification: Not sure, haven't seen that before. Anyone? This comment refers to: Looks like the PBX isn't sending the SIP messages- I notice the previous NOTIFY messages said (queued)- does this mean anything? Philipp -- _ -- 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] MOH Oddity
On Fri, Mar 5, 2010 at 11:29 PM, Matt mhop...@gmail.com wrote: I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. If the MOH files have the same codec as the calls, I don't really think you need to think of this as resource demanding. Normal MOH from files only opens the files and reads a chunk of data, and if possible sends it directly to the client. This means that there is no advanced player in use. If your machine have reasonable amount of ram installed, the files also will automatically be cached in your ram by your operating system. So it's really just transactions of data chunks from your ram to a socket. Doesn't really matter if all channels read from the same place inside the file, or if they start from the beginning for each channel. If you used a single external program to convert for example a web radio to 8khz audio, then you would get all those calls retrieving data from the same place. But this would be much more prone to errors, and more resource demanding than pre-encoded audio files in a directory like you already do. Håkon -- _ -- 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] MOH Oddity
For some reason I have to set the type to 'files' if I set it to 'quietmp3' I get nothing, even though the files are valid MP3 files that play on another asterisk system... does that mean I've got something installed wrong? 2010/3/5 Håkon Nessjøen haa...@avelia.no On Fri, Mar 5, 2010 at 11:29 PM, Matt mhop...@gmail.com wrote: I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as I might, I can not get it to work. If the MOH files have the same codec as the calls, I don't really think you need to think of this as resource demanding. Normal MOH from files only opens the files and reads a chunk of data, and if possible sends it directly to the client. This means that there is no advanced player in use. If your machine have reasonable amount of ram installed, the files also will automatically be cached in your ram by your operating system. So it's really just transactions of data chunks from your ram to a socket. Doesn't really matter if all channels read from the same place inside the file, or if they start from the beginning for each channel. If you used a single external program to convert for example a web radio to 8khz audio, then you would get all those calls retrieving data from the same place. But this would be much more prone to errors, and more resource demanding than pre-encoded audio files in a directory like you already do. Håkon -- _ -- 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 -- _ -- 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] MOH Oddity
Un-top-posting... On Fri, Mar 5, 2010 at 11:29 PM, Matt mhop...@gmail.com wrote: I'm trying to setup my asterisk system for the least overhead as possible. On Fri, 5 Mar 2010, Matt wrote: For some reason I have to set the type to 'files' if I set it to 'quietmp3' I get nothing, even though the files are valid MP3 files that play on another asterisk system... does that mean I've got something installed wrong? Least overhead as possible means don't use MP3 files. If you have so much MOH activity to warrant your attention, you should use files encoded with the codec used by the channel. Or, you should hire more agents :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] MOH Oddity
On Friday 05 March 2010 17:19:06 Matt wrote: For some reason I have to set the type to 'files' if I set it to 'quietmp3' I get nothing, even though the files are valid MP3 files that play on another asterisk system... does that mean I've got something installed wrong? Mostly likely you didn't install mpg123. 'Files' mode starts a separate stream for each user, while mp3 and quietmp3 modes use mpg123 to share a single stream amongst all listeners. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
