Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011 17:35:31, Asterisk Security Team a écrit :
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SusceptibilityRemote Authenticated Sessions
Severity Moderate
Exploits KnownNo
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated OnJanuary 18, 2011
Advisory Contact Matthew Nicholson mnichol...@digium.com
CVE Name
Description When forming an outgoing SIP request while in pedantic mode, a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well.
The ast_uri_encode function does not properly respect the size
of its output buffer and can write past the end of it when
encoding URIs.
Resolution The size of the output buffer passed to the ast_uri_encode
function is now properly respected.
In asterisk versions not containing the fix for this issue,
limiting strings originating from remote sources that will be
URI encoded to a length of 40 characters will protect against
this vulnerability.
exten = s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
exten = s,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten = s,n,Dial(SIP/channel)
The CALLERID(num) and CALLERID(name) channel values, and any
strings passed to the URIENCODE dialplan function should be
limited in this manner.
Affected Versions
Product Release Series
Asterisk Open Source1.2.x All versions
Asterisk Open Source1.4.x All versions
Asterisk Open Source1.6.x All versions
Asterisk Open Source1.8.x All versions
Asterisk Business Edition C.x.x All versions
AsteriskNOW 1.5 All versions
s800i (Asterisk Appliance) 1.2.x All versions
Corrected In
Product Release
Asterisk Open Source 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1,
1.6.2.16.1, 1.8.1.2, 1.8.2.1
Asterisk Business Edition C.3.6.2
Patches
URL Branch
http://downloads.asterisk.org/pub/security/AST-2011-001-1.4.diff1.4
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.1.diff 1.6.1
http://downloads.asterisk.org/pub/security/AST-2011-001-1.6.2.diff 1.6.2
http://downloads.asterisk.org/pub/security/AST-2011-001-1.8.diff1.8
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security
This document may be superseded by later versions; if so, the latest
[asterisk-users] How to detect line tone?
I need in a strange applicatio a way to detect the tone (busy, ring etc. etc.) of analog line (zap channel), while channel UP. I found the application NV line detect, but is very old, and may be not mantained. I can patch asterisk to actually support this application but i think someone other have something like this done. Thnks. attachment: massimo.vcf signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.2 and digium yum repositories
Hi Does anyone have any idea how long it will take for the new release of asterisk 1.8 to make it to the digium yum repositories? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Calling rules
Correcting the line to:
exten =
_0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
problem persists...
any other suggestions?
Best regards,
What does your trunkdial-failover-0.3 look like?
Here goes...
[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} 6]?1-fmsetcid,1)
exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} 1]?1-setgbobname,1)
exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})}
2]?${CID_${CALLERID(num)}}:)})
exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} 6]?1-dial,1)
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})}
6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})})
exten = s,n,Goto(1-dial,1)
exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
exten = 1-setgbobname,n,Goto(s,3)
exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
exten = 1-fmsetcid,n,Goto(1-dial,1)
exten = 1-dial,1,Dial(${ARG1})
exten = 1-dial,n,Gotoif(${LEN(${ARG2})} 0 ?1-${DIALSTATUS},1:1-out,1)
exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
exten = 1-CHANUNAVAIL,n,Hangup()
exten = 1-CONGESTION,1,Dial(${ARG2})
exten = 1-CONGESTION,n,Hangup()
exten = 1-out,1,Hangup()
--
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[asterisk-users] Asterisk fail over. From IP rewrite issues
Hey guys, I hope somebody has some experience with the following because i'm stuck ;-). I'm creating a fail over situation for Asterisk and this works great. The only issue i have so fair os the from ip. I used the IP fix routing here - http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions and that works, in it's own subnet. But when the Asterisk box is in 10.100.2.x and my phones are in 10.100.3.x i have no audio. (All the other stuff seems to work fine.. i can auto provision, make the call (the phone will ring). When i check the SIP debugging i see a correct from IP. I think that the issue is with this line: ip route change 10.10.10.0/24 src 10.10.10.110 dev eth0 And it sould be: ip route change 10.10.0.0/16 src 10.10.10.110 dev eth0 But when i try to enable that line i get a error saying that the thing i'm trying to change is not there. Anybody got any input on this issue? Would be great! Thanks, 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] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them
Hello list, what does this mean in the debug-log : [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Read factory 0x153cec40 and write factory 0x153cf678 both fail to provide 160 samples [Jan 19 15:11:04] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them. [Jan 19 15:11:05] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:05] DEBUG[1475] audiohook.c: Read factory 0x153cec40 and write factory 0x153cf678 both fail to provide 160 samples [Jan 19 15:11:05] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:05] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them. [Jan 19 15:11:05] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:05] DEBUG[1475] audiohook.c: Read factory 0x153cec40 and write factory 0x153cf678 both fail to provide 160 samples My debug file is flooded with this message... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote: When you get over 500 emails a day on your blackberry you have make a decision on what is or is not worth reading at that moment. Its not lazy at all its cutting through the fluff and finding the emails worth while. When inside outlook you don't have the hot key b to scroll to the bottom so again, I'd have to scroll down. Add up the time it takes per email x 500 emails, you loose considerable amount of productivity. Oh C'mon this is definitely lazy never heard of CTRL+END it works in Outlook Top posting has its useful place as well as bottom posting. Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Fred Posner f...@teamforrest.com Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 16 Jan 2011 21:43:00 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On Mon, 2011-01-17 at 02:31 +, James Miller wrote: I hate to disagree but I find it much, much easier to follow conversations when the newest reply is on top. I find it too time consuming to scroll through a long message just to find out someone left a three word reply. As I am on my blackberry more than I am at a pc, if I don't see the reply as soon as I open the message it gets deleted without being read. Time is money and I don't have time to scroll through every message. I will agree that sometimes it is helpful to make replies at the bottom and I will attempt to keep the peace by posting at the bottom when I can, but top posting is easier and more clean to read than having 100 lines of and broken lines. Warmest regards, James Sent from my Verizon BlackBerry. Always on, Always Connected -Original Message- From: Lesly Dorval lador...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 17 Jan 2011 02:14:54 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting Shaun Ruffell sruffell at digium.com writes: Whatever your preferred style, the following post is at least worth considering. http://brooksreview.net/2011/01/interleaved-email/ My belief is that it would be nearly impossible for me to follow a high volume list if top posting was the preferred style. For example, the following email from the LKML would need to be more verbose if all the participants were top posting, because they would all have to set the context for their comments. Instead, you can follow the chain of thought for each of the threads contained in the email. http://article.gmane.org/gmane.linux.kernel/1087665 Anyway, just something to consider, Shaun I could never understand the strong objection regarding top-posting until Shaun shared these examples - though I had been reading lists for more years than I care to admit. These examples clearly show how snipping and bottom posting translate to susccint and clear contextual communication. From now I will evangelize snipping and bottom posting. I cannot imagine considering scrolling to the end of an email time consuming. Very sad. If you find it too difficult on your blackberry to press the B key (to jump to the bottom of the message) then I am uncertain how you have enough time to even read this email. I'm all for good arguments. That time consuming one is just lazy. I personally find top posting annoying and only serving to an immediate conversation. Particularly useless if referencing the message later. -- With best regards, ---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 -- _ -- 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:
[asterisk-users] agi dial termination cause ?
Hi All, in an AGI script, if executing the Asterisk command Dial, I only get result = -1 (if the call has been answered by the callee) and result = 0 (for everything else) Question: how can I know if the call was not answered because of timeout or because the callee was busy ? (I'm using Asterisk 1.8) Thank you very much for supporting, regards and have a nice day. Mike -- _ -- 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] agi dial termination cause ?
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because the callee was busy ?
(I'm using Asterisk 1.8)
Thank you very much for supporting,
regards and have a nice day.
Mike
--
Take a look here:
http://www.voip-info.org/wiki/view/Asterisk+variables
Perhaps you can get this info from ${ANSWEREDTIME} or from
${DIALSTATUS}.
-Thorsten-
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Re: [asterisk-users] agi dial termination cause ?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Wednesday, January 19, 2011 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi dial termination cause ?
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because
the callee was busy ?
(I'm using Asterisk 1.8)
Thank you very much for supporting,
regards and have a nice day.
Mike
--
Take a look here:
http://www.voip-info.org/wiki/view/Asterisk+variables
Perhaps you can get this info from ${ANSWEREDTIME} or from ${DIALSTATUS}.
-Thorsten-
You may end up needing to use a context to dial and get the desired results
instead of using the dial command.
--
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Re: [asterisk-users] agi dial termination cause ?
On Wed, 19 Jan 2011 17:03:03 +0100
Thorsten Göllner t...@ovm-group.com wrote:
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because the
callee was busy ?
(I'm using Asterisk 1.8)
Thank you very much for supporting,
regards and have a nice day.
Mike
--
Take a look here:
http://www.voip-info.org/wiki/view/Asterisk+variables
Perhaps you can get this info from ${ANSWEREDTIME} or from ${DIALSTATUS}.
-Thorsten-
Ohh great! I have forgot about them, thank you both very much!
I confirm that if using phpagi the array $agi-get_variable(DIALSTATUS)
['data'] gets populated with NOANSWER, BUSY, CANCEL, ...
Thank you again and have a nice day :)
Mike
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Re: [asterisk-users] Top Posting
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful. :)
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Re: [asterisk-users] Asterisk 1.8.2 and digium yum repositories
On 01/19/2011 04:41 AM, Ishfaq Malik wrote: Hi Does anyone have any idea how long it will take for the new release of asterisk 1.8 to make it to the digium yum repositories? Thanks Ish They've been there since yesterday afternoon. It's possible that you hit the repository before the packages were there, causing the refresh timer to be extended (the default is probably 24 hours - but I'd have to check). If they still aren't showing up for you, you can run `yum clean metadata; yum update` -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
11:39 Parker said
That would fall under Quirk's Exception: Intentionally invoking Godwin's
Law to attempt to kill a thread is rarely successful. :)
Didn't work this time :)
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[asterisk-users] intermittent problem on 1.4
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline-voipfone-asterisk 1.4-voipfone-UK landline About 1 in 3 times the call at the final landline is silent and we see RTP Read too short scrolling on the console log. Where do we start working out what's going on? Other than that the server is working well 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] intermittent problem on 1.4
John Taylor wrote: [snip] Where do we start working out what's going on? Other than that the server is working well John could you please ilustrate a little bit more your scenario?, (if you want, use fake IPs). Note: What's the exactly version number of your Asterisk box? -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] IAX between 1.6 and 1.8 has bad voice quality
I recently upgraded my office server to 1.8 and since then I have very bad voice quality when calling another Asterisk server that uses 1.6. The links is via IAX2 and I have tried using g729 and ulaw but I still have the same problem although ulaw has a slight better result. Any changes that need to me made to the IAX2 trunk for it to work? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Wed, Jan 19, 2011 at 6:47 PM, Don Kelly d...@donkelly.biz wrote: 11:39 Parker said That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to attempt to kill a thread is rarely successful. :) Didn't work this time :) Slightly OT: why is the Gmail ad server, which is usually all about PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on this thread? Are they seeing it as that childish? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Wed, Jan 19, 2011 at 2:37 PM, randulo rand...@randulo.com wrote: Slightly OT: why is the Gmail ad server, which is usually all about PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on this thread? Are they seeing it as that childish? /r Also OT: Google combines message context with your personal search history to do ad targeting, so look in the mirror. I just made that up, though. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen Sent: Wednesday, January 19, 2011 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Wed, Jan 19, 2011 at 2:37 PM, randulo rand...@randulo.com wrote: Slightly OT: why is the Gmail ad server, which is usually all about PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on this thread? Are they seeing it as that childish? /r Also OT: Google combines message context with your personal search history to do ad targeting, so look in the mirror. I just made that up, though. -M Not your mirror - your cookies! -- _ -- 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 extension not found problem...
Hi Steve, The asterisk CLI shows the context of caller as below: *moment-portable*CLI sip show user IMSI310410270465840 moment-portable*CLI * Name : IMSI310410270465840 Secret : Not set MD5Secret: Not set Context : sip-external Language : AMA flags: Unknown Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : 2102 ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs Codec Order : (gsm:20) Auto-Framing: No *But when I do dialplan show 2103@sip-external, it returns no dialplan *moment-portable*CLI dialplan show 2103@sip-external There is no existence of 'sip-external' context Command 'dialplan show 2103@sip-external' failed. * I have already created a dialplan in my extensions.conf, I am not sure what is happening here ?? Badly need help in this. Thanks, Abhinav On Tue, Jan 18, 2011 at 8:37 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 18 Jan 2011, abhinav anand wrote: The exact error thrown on Asterisk CLI is chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to extension 2103 rejected because extension not found What context does 'sip show user IMSI310410270465840' show? What does 'dialplan show 2103@context-from-previous-command' show? -- 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] Asterisk extension not found problem...
On Wed, 19 Jan 2011, abhinav anand wrote: The asterisk CLI shows the context of caller as below: moment-portable*CLI sip show user IMSI310410270465840 Context : sip-external But when I do dialplan show 2103@sip-external, it returns no dialplan moment-portable*CLI dialplan show 2103@sip-external There is no existence of 'sip-external' context Command 'dialplan show 2103@sip-external' failed. I have already created a dialplan in my extensions.conf, I am not sure what is happening here ?? 1) Do you need to do a 'dialplan reload?' 2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read? 3) Do you start Asterisk with the ? command line option? 4) What is the value of 'astetcdir' in asterisk.conf? -- 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] Asterisk extension not found problem...
On Wed, 19 Jan 2011, Steve Edwards wrote: 3) Do you start Asterisk with the ? command line option? ? = '-C' -- 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] Asterisk extension not found problem...
Hi Steve, Here are the answers to the questions. *1) Do you need to do a 'dialplan reload?'* I don't need to do a dialplan reload. Infact there is no such command as dialplan reload. I simply do a reload each time I make a config change. *2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read?* There are two extensions.conf files present in */etc/asterisk/extensions.conf /home/moment/openbts-uhd/public-trunk/AsteriskConfig/extensions.conf *I am making the changes in /etc/asterisk file. However, when I have tried putting same changes in other file too but again no success. *3) Do you start Asterisk with the ? command line option?* I start Asteisk using sudo asterisk -vvvgcr or sudo asterisk -r. *-c says Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect*. . *4) What is the value of 'astetcdir' in asterisk.conf?* The value is as astetcdir = /etc/asterisk and other values are: [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /usr/share/asterisk astagidir = /usr/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Some extra information: - My asterisk version is *Asterisk 1.6.2.5-0ubuntu1.1 built by buildd @ palmer on a i686 running Linux on 2010-07-16 13:24:33 UTC* - I am not able to verify the symlink between the two extensions.conf files Thanks, Abhinav On Wed, Jan 19, 2011 at 2:38 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 19 Jan 2011, abhinav anand wrote: The asterisk CLI shows the context of caller as below: moment-portable*CLI sip show user IMSI310410270465840 Context : sip-external But when I do dialplan show 2103@sip-external, it returns no dialplan moment-portable*CLI dialplan show 2103@sip-external There is no existence of 'sip-external' context Command 'dialplan show 2103@sip-external' failed. I have already created a dialplan in my extensions.conf, I am not sure what is happening here ?? 1) Do you need to do a 'dialplan reload?' 2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read? 3) Do you start Asterisk with the ? command line option? 4) What is the value of 'astetcdir' in asterisk.conf? -- 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] Cross Queue Priorities
Morning All, My Google skills may be failing me as I can see people asking this but no useful responses, I need a way to prioritise calls across queues - I can think of ways to do this but they are far from elegant and this seems like such a simple request I am sure I am missing something obvious. All my queues are of equal weight (Ie. A caller in Queue A can be just as important as a caller in Queue B) but not all my callers are of equal priority - Ie. A caller in Queue A with a priority of 100 needs to reach an agent before a priorty 50 call in Queue B, keeping in mind that a single agent can be in both Queue A and Queue B. Would appreciate any input on this at all! Cheers Nick. -- _ -- 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 extension not found problem...
Please do not add me or yourself to the address list. We should keep the discussion on the list (and just the list) so it is available to everyone. Also, top-posting is 'frowned upon.' On Wed, 19 Jan 2011, abhinav anand wrote: Here are the answers to the questions. 1) Do you need to do a 'dialplan reload?' I don't need to do a dialplan reload. Infact there is no such command as dialplan reload. I simply do a reload each time I make a config change. What version of Asterisk are you using? 1.2 = 'extensions reload' 1.6 = 'dialplan reload' (I don't have a 1.4 or 1.8 on hand.) If you don't have one of these, something is seriously wrong. 2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read? There are two extensions.conf files present in /etc/asterisk/extensions.conf /home/moment/openbts-uhd/public-trunk/AsteriskConfig/extensions.conf I am making the changes in /etc/asterisk file. However, when I have tried putting same changes in other file too but again no success. 3) Do you start Asterisk with the -C command line option? I start Asteisk using sudo asterisk -vvvgcr or sudo asterisk -r. -c says Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. The 'r' command line option asks to connect to an existing instance, so this is not the command you use to start Asterisk. The 'upper-case C' command line option allows you to specify location other than /etc/asterisk/ for asterisk.conf. Typing 'echo $(cat /proc/pid-of-asterisk/cmdline)' will show the command line and options Asterisk was started with. 4) What is the value of 'astetcdir' in asterisk.conf? The value is as astetcdir = /etc/asterisk and other values are: [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /usr/share/asterisk astagidir = /usr/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Some extra information: - My asterisk version is Asterisk 1.6.2.5-0ubuntu1.1 built by buildd @ palmer on a i686 running Linux on 2010-07-16 13:24:33 UTC So 'dialplan reload' would be the proper command to just reload the dialplan. - I am not able to verify the symlink between the two extensions.conf files If you edit one and your edits don't magically appear in the other, they are not linked. The 'ls' command can also be use to confirm 'linkness.' When you do a 'dialplan show' do you see lines like: 1. mumble-mumble [pbx_config] or 1. mumble-mumble [pbx_ael] or both? (pbx_config means extensions.conf, pbx_ael means extensions.ael) -- 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] Asterisk extension not found problem...
Hi Steve,
I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan
reload. When I do core show help dialplan I get list of commands as:
*
moment-portable*CLI core show help dialplan
dialplan debug Show fast extension pattern matching data
structures
dialplan set chanvar Set a channel variable
dialplan set extenpatternmatch Use the Old extension pattern
matching algorithm.
dialplan set extenpatternmatch Use the New extension pattern
matching algorithm.
dialplan set global Set global dialplan variable
dialplan show chanvar Show channel variables
dialplan show globals Show global dialplan variables
dialplan show Show dialplan
*
Also executing echo $(cat /proc/pid-of-asteriskcmdline) returned this
path
*/usr/sbin/asterisk-p-Uasterisk*
I have verified the symlink between two extensions.conf files. It is okay
now.
My dialplan show returns some 28 contexts (*all pbx_ael and no pbx_config*)
and looks like this
(seems context are read from extensions.ael file only)
*[ Context 'default' created by 'pbx_lua' ]
Alt. Switch ='Lua/'[pbx_lua]
moment-portable*CLI
[ Context 'demo' created by 'pbx_lua' ]
Alt. Switch ='Lua/'[pbx_lua]
moment-portable*CLI
[ Context 'local' created by 'pbx_lua' ]
Alt. Switch ='Lua/'[pbx_lua]
moment-portable*CLI
[ Context 'ael-default' created by 'pbx_ael' ]
Include ='ael-demo'[pbx_ael]
[ Context 'ael-demo' created by 'pbx_ael' ]
'#' =1. Playback(demo-thanks) [pbx_ael]
2. Hangup() [pbx_ael]
'1000' = 1. Goto(ael-default,s,1) [pbx_ael]
'2' =1. Background(demo-moreinfo) [pbx_ael]
2. Goto(s,instructions) [pbx_ael]
'3' =1. Set(LANGUAGE()=fr) [pbx_ael]
2. Goto(s,restart)[pbx_ael]
'500' = 1. Playback(demo-abouttotry) [pbx_ael]
2. Dial(IAX2/gu...@misery.digium.com/s@default)
[pbx_ael]
3. Playback(demo-nogo)[pbx_ael]
4. Goto(s,instructions) [pbx_ael]
'600' = 1. Playback(demo-echotest)[pbx_ael]
2. Echo() [pbx_ael]
3. Playback(demo-echodone)[pbx_ael]
4. Goto(s,instructions) [pbx_ael]
'8500' = 1. VoicemailMain()[pbx_ael]
2. Goto(s,instructions) [pbx_ael]
'i' =1. Playback(invalid) [pbx_ael]
's' =1. Wait(1)[pbx_ael]
2. Answer() [pbx_ael]
3. Set(TIMEOUT(digit)=5) [pbx_ael]
4. Set(TIMEOUT(response)=10) [pbx_ael]
[restart] 5. Background(demo-congrats) [pbx_ael]
[instructions] 6. MSet(x=$[0]) [pbx_ael]
7. GotoIf($[ ${x} 3]?8:12) [pbx_ael]
8. Background(demo-instruct) [pbx_ael]
9. WaitExten()[pbx_ael]
10. MSet(x=$[${x} + 1]) [pbx_ael]
11. Goto(7) [pbx_ael]
12. NoOp(Finish for-ael-demo-3) [pbx_ael]
't' =1. Goto(#,1) [pbx_ael]
'_1234' =1. Gosub(ael-std-exten-ael,s,1(${EXTEN}, IAX2))
[pbx_ael]
[ Context 'ael-local' created by 'pbx_ael' ]
Include ='ael-default' [pbx_ael]
Include ='ael-trunklocal' [pbx_ael]
Include ='ael-iaxtel700' [pbx_ael]
Include ='ael-trunktollfree' [pbx_ael]
Include ='ael-iaxprovider' [pbx_ael]
Ignore pattern = '9' [pbx_ael]
[ Context 'ael-longdistance' created by 'pbx_ael' ]
Include ='ael-local' [pbx_ael]
Include ='ael-trunkld' [pbx_ael]
Ignore pattern = '9' [pbx_ael]
[ Context 'ael-international' created by 'pbx_ael' ]
Include ='ael-longdistance'
Re: [asterisk-users] Asterisk extension not found problem...
Un-top-posting... On Wed, 19 Jan 2011, abhinav anand wrote: I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan reload. If you do not have 'dialplan reload,' you do not have pbx_config.so loaded. Since pbx_config.so reads extensions.conf, if you don't have it loaded, extensions.conf will not be read. My dialplan show returns some 28 contexts (all pbx_ael and no pbx_config) and looks like this (seems context are read from extensions.ael file only) So, you need to either load pbx_config.so to read your extensions.conf or add the 'sip-external' context to extensions.ael. -- 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] Asterisk extension not found problem...
On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote: Un-top-posting... On Wed, 19 Jan 2011, abhinav anand wrote: I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan reload. If you do not have 'dialplan reload,' you do not have pbx_config.so loaded. Since pbx_config.so reads extensions.conf, if you don't have it loaded, extensions.conf will not be read. My dialplan show returns some 28 contexts (all pbx_ael and no pbx_config) and looks like this (seems context are read from extensions.ael file only) So, you need to either load pbx_config.so to read your extensions.conf or add the 'sip-external' context to extensions.ael. The last time this happened to me was because extensions.conf had some strange characters that prevented it from loading. Also check that you are not trying to load it from a database by means of extconfig.conf -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 extension not found problem...
Thanks Steve, I figured out the problem. As you said correctly, *pbx_config.so* was not getting loaded because in my extensions.conf file one extra file extensions.local.conf was included which was actually not present in the directory. I have commented that line and did *module load pbx_config.so* to reload pbx_config.so and now I see both dialplan reload and my sip-external extensions correctly. many many thanks to you for all your pointers and input. I hope this resolves my issue. Thanks to Carlos too for pointing out the reason. Fortunately I am saved from extconfig.conf thing :) Thanks, Abhinav On Wed, Jan 19, 2011 at 4:43 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote: Un-top-posting... On Wed, 19 Jan 2011, abhinav anand wrote: I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan reload. If you do not have 'dialplan reload,' you do not have pbx_config.so loaded. Since pbx_config.so reads extensions.conf, if you don't have it loaded, extensions.conf will not be read. My dialplan show returns some 28 contexts (all pbx_ael and no pbx_config) and looks like this (seems context are read from extensions.ael file only) So, you need to either load pbx_config.so to read your extensions.conf or add the 'sip-external' context to extensions.ael. The last time this happened to me was because extensions.conf had some strange characters that prevented it from loading. Also check that you are not trying to load it from a database by means of extconfig.conf -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] No more ISDN in Malaysia Telekom???
We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- 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 extension not found problem...
On Wed, 19 Jan 2011, abhinav anand wrote: I figured out the problem. As you said correctly, pbx_config.so was not getting loaded because in my extensions.conf file one extra file extensions.local.conf was included which was actually not present in the directory. I have commented that line and did module load pbx_config.so to reload pbx_config.so and now I see both dialplan reload and my sip-external extensions correctly. No. A missing '#include' file will not cause pbx_config.so to fail to load. pbx_config.so has to be loaded since IT reads extensions.conf and any subsequent included files. Similarly, garbage in extensions.conf does not affect the loading of pbx_config.so because it is already loaded. If pbx_config.so is not being loaded, look at modules.conf for clues -- specifically, autoload and noload. Since you loaded pbx_config.so 'by hand,' it will 'go away' the next time Asterisk is restarted. -- 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] 1.8.2: dahdi-2.4: calls dropping
On 01/18/2011 08:17 PM, Shaun Ruffell wrote: On 1/18/11 6:55 PM, sean darcy wrote: On 01/18/2011 05:27 PM, Shaun Ruffell wrote: On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack .. -- Executing [s@incoming-pstn-line:6] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 6) exited non-zero on 'DAHDI/4-1' -- Hanging up on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives? sean Just a WAG - the bridge isn't really happening and you're getting a dial timeout. If you were running trunk...this is a very good guess. The following commit resolved an issue with bridging that's been in trunk for the past few weeks. http://svn.asterisk.org/view/dahdi?view=revisionrevision=9642 Wasn't running trunk. It was the 1.8.2 release. Not sure I understand: the dial timeout is 36 seconds. Yet the call doesn't drop for at least 5, probably 10, maybe more minutes. And no audio was muted while the call was up. It was all just fine. What card are you using to access the PSTN. It's possible there might be some debug flags you can enable to see if the board thinks the FXS port is flashing. Is this a new installation or are you suddenly having this problem on an old installation? dahdi_hardware Unrecognized garbage 'Reserved' in WCTDM/4/2 pci::01:05.0 wctdm+ e159:0001 Wildcard TDM400P REV I This installation is 3, maybe 4 years old. Thanks for trying to help. 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] No more ISDN in Malaysia Telekom???
Hello Lee, Telekom Malaysia provide PRI lines. We've been actively using their services for the past few years with success. Let me know if you need contacts. Regards, Arstan On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney) john@compuware.comwrote: We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- 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] No more ISDN in Malaysia Telekom???
Arstan, thank you for your response. Malaysia Telekom replied This service is limited to avaibility of ports and infra avaibility as we are now upgrading to NGN. You may use business broadband and PSTN lines to connect to your Digital PABX to replace this service. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan Jusupov Sent: Thursday, 20 January 2011 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No more ISDN in Malaysia Telekom??? Hello Lee, Telekom Malaysia provide PRI lines. We've been actively using their services for the past few years with success. Let me know if you need contacts. Regards, Arstan On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney) john@compuware.com wrote: We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- 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] res_fax
On Jan 19, 2011, at 3:18 PM, Jason Parker wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Am I the only one confused here? (probably) It seems like you imply that SendeFax (which looks like a typo to me) is correct in the second sentence, then reverse yourself in the last parenthetical statement. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Jan 19, 2011, at 10:06 AM, C F wrote: On Sun, Jan 16, 2011 at 9:47 PM, James Miller paramedi...@gmail.com wrote: When you get over 500 emails a day on your blackberry you have make a decision on what is or is not worth reading at that moment. Its not lazy at all its cutting through the fluff and finding the emails worth while. When inside outlook you don't have the hot key b to scroll to the bottom so again, I'd have to scroll down. Add up the time it takes per email x 500 emails, you loose considerable amount of productivity. Oh C'mon this is definitely lazy never heard of CTRL+END it works in Outlook How amusing that you follow that statement by being too lazy to trim all of the irrelevant crud after your comment by pressing ctrl-shift-end followed by delete. It works in Outlook. Tom -- _ -- 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] Calling rules
On Jan 19, 2011, at 5:06 AM, Vitor Carlos Flausino wrote: In other words, which of the following is your situation: 1.) User dials 0X, asterisk sends 0X to the telco. 2.) User dials 0X, asterisk parses 0, strips it, and sends X to the telco. That might narrow it down. Option 2. 0 is to get an external line and XXX is passed to telco. -vcf It seems to me that you are passing the 0 to the telco when the user dials all digits at once. When they dial the 0 first, the call gets sent to one extension (probably extension 0 or _0) and just connects them to the outside line, sending nothing to the telco. When they dial 0X, asterisk matches another extension (probably _0. or another that begins with _0), one that connects them to the outside line and sends everything out to the telco, including the 0. Just a guess, but it sounds right to me. If so, you need to modify the dial command to strip the 0 before sending it. Tom -- _ -- 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] Internode weirdness
I have an updated asterisk 1.8 server running on Freebsd 8.1, and connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl connection (in other words FreeBSD is doing all the hard work). I am trying to connect with Internode nodephone, but they aren't really willing to spend the time to work it out (depending on who you get to talk to), and they reckon its all working as it should. I was originally running a 1.4 server trying to get it working, but when I didn't have a great deal of success setting it up, and I noticed features were missing that I wanted, and 1.8 was finally ported, I jumped on the chance and updated. I was originally able to get outgoing calls working after quite a bit of fiddling with settings, but no incoming. I finally found some info to tweak the firewall to suit the asterisk and voip services, and now I can finally get perfect incoming calls- but now outgoing won't work at all! :( I've been hammering at this for days now- working my google foo like crazy to get some clues as to why. Nada... So what am I missing? The only facts I have are: Internode insist their setup gets around NAT issues, so in an ordinary ATA setup you don't need nat. The proviso is that it needs to be on a dmz- basically they say open all connections from their server and direct them to the ATA. (I did have outgoing calls working in this scenario, but I couldn't get incoming; and to boot if I had other clients outside the NAT- which I am looking at doing as well, just not going through internode- it basically won't allow it) The firewall is setup to NAT port 5060 as 5060 to the internode server and redirected on return. RTP 1-2 is directed through to the server as well. SIP debug on: On making an outgoing call I get retransmission timeout errors and this: WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up. The dialog can change too- if I change fromdomain it changes accordingly. -- SIP/sip-out-001d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Tcpdumps and logs show messages going out of asterisk and both interfaces on the firewall, but none coming in. Registry and peers list show the Internode connections are fine- qualifying is working. I have also followed recommendations to separate incoming and outgoing peers (despite the added complexity), so I have an sip-in and sip-out peers with settings for internode; although even if I comment out one and adjust the dialplan it still shows the same error. I also tried turning off the externip setting- no luck. I'm at the end of my tether- I'm ready to turn a laptop into a missile! And the lack of interest is killing me Any help would be much appreciated at this point- its doing my head in! Cheers -- _ -- 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] res_fax
There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Am I the only one confused here? (probably) It seems like you imply that SendeFax (which looks like a typo to me) is correct in the second sentence, then reverse yourself in the last parenthetical statement. I'm not confused if he means that the content of Application_SendeFax is correct and the content of Application_SendFax is old. After the next update, the content of Application_SendFax will be correct and Application_SendeFax will go away --Don -- _ -- 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] Hi, agent intro-speech for outside caller
Hello, I'm using AsteriskNow. Asterisk version is 1.6.2.15 and FreePBX 2.7.0.0 Is there anyway to play prerecorded agent intro-speech (like Hello, my name is ) to outside caller when agent picks up? thank you -- _ -- 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] No more ISDN in Malaysia Telekom???
Hi Lee, yes, it depends on the location. Usually they will check the location to see if it is available there. Do you have your location set already? If you need help further help, we can take our conversation off the mailing list. Arstan On Thu, Jan 20, 2011 at 11:14 AM, Lee, John (Sydney) john@compuware.com wrote: Arstan, thank you for your response. Malaysia Telekom replied This service is limited to avaibility of ports and infra avaibility as we are now upgrading to NGN. You may use business broadband and PSTN lines to connect to your Digital PABX to replace this service. From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan Jusupov Sent: Thursday, 20 January 2011 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No more ISDN in Malaysia Telekom??? Hello Lee, Telekom Malaysia provide PRI lines. We've been actively using their services for the past few years with success. Let me know if you need contacts. Regards, Arstan On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney) john@compuware.com wrote: We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- 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 -- _ -- 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] Using asterisk and icecast for live audio streaming.
Hi all, Can someone give me a direction on how to use asterisk and icecast or any other apps for a live audio cast? The audio feed is external to the asterisk server. Voip-info.org is not detailed on this. Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Also OT: Google combines message context with your personal search history to do ad targeting, so look in the mirror. I just made that up, though. Not your mirror - your cookies! No, it's true! Now I'm seeing Untimate Black Hat SEO (yes misspelled because Ultimate was too expensive) I was just looking at an SEO report site about top posting and they say lists.digium.com is number 1 and needs no help. And I do kind of look like Justin Beiber will in about a half-century from now. That's why I have broken all the mirrors in the house. /r -- _ -- 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
