Re: [asterisk-users] Asterisk 1.8.2 and digium yum repositories
On Wed, 2011-01-19 at 11:41 -0600, Jason Parker wrote: 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` Thanks a lot, whereas I've been playing with asterisk for a few years now I've only been playing with CentOS for 2 weeks. I'll remember that tip for the future. -- 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] Top Posting
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 This is the problem, everyone has a personal goal. One side wants fast replies at the top, with no interest in the repetitive, redundant signature/disclaimers content below. The other wants total historical readability or questions and answers in top to bottom readability in every message. And, this is a type of list that is used by 1000s of individuals, not people from a single company. We are just lucky we don't have someone posting in sentences that read from right to left. :-) Also most (all?) mail clients don't allow setting preferences based on the source of the message. I.E. Top post for email, bottom post for the cooking list and bottom post for the Asterisk list. And then almost no one trims anything no matter what their preferences/beliefs are, and yells at others for top or bottom posting or interleaving, usually while violating some other list rule or general net etiquette. How about just no quoting or only the actual last message you are replying to? The list doesn't require any quoting. Contribute your thoughts, and leave it at that. Everyone has the previous posts on their computer, if they don't know the history, let them go back and read. Cary -- _ -- 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] Hi, agent intro-speech for outside caller
On Jan 19, 2011, at 11:08 PM, DSR wrote: Is there anyway to play prerecorded agent intro-speech (like Hello, my name is ) to outside caller when agent picks up? I don't know of a way to do that, but I can say that, as a caller, it is highly annoying. Your agents ought to be able to do that themselves, no? -- _ -- 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] Hi, agent intro-speech for outside caller
On 20 January 2011 11:54, Tom Rymes try...@rymes.com wrote: I don't know of a way to do that, but I can say that, as a caller, it is highly annoying. Your agents ought to be able to do that themselves, no? Exactly, otherwise you are losing first chance to make the call different from the other ones where caller feel like they are talking with machines. Simple Hello, it's X, how is your day today sir (and given it's a bit different every day) can change they way the call is going to go... -- _ -- 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 Cary Fitch Sent: Thursday, January 20, 2011 3:06 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Top Posting 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 This is the problem, everyone has a personal goal. One side wants fast replies at the top, with no interest in the repetitive, redundant signature/disclaimers content below. The other wants total historical readability or questions and answers in top to bottom readability in every message. And, this is a type of list that is used by 1000s of individuals, not people from a single company. We are just lucky we don't have someone posting in sentences that read from right to left. :-) Also most (all?) mail clients don't allow setting preferences based on the source of the message. I.E. Top post for email, bottom post for the cooking list and bottom post for the Asterisk list. And then almost no one trims anything no matter what their preferences/beliefs are, and yells at others for top or bottom posting or interleaving, usually while violating some other list rule or general net etiquette. How about just no quoting or only the actual last message you are replying to? The list doesn't require any quoting. Contribute your thoughts, and leave it at that. Everyone has the previous posts on their computer, if they don't know the history, let them go back and read. Cary Possibly the most literate and civil post in this flame-war... Two points to add - #1 if you don't have the history on your computer, the nice folks at Asterisk/Digium keep all of this online for posterity #2 It's definitely not a good idea to keep the entire thread intact since the server at A/D holds the message once it exceeds 40K. No matter what your posting posture do everyone a favor and trim before replying... -- _ -- 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] Hi, agent intro-speech for outside caller
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Swedrowski Sent: Thursday, January 20, 2011 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hi, agent intro-speech for outside caller On 20 January 2011 11:54, Tom Rymes try...@rymes.com wrote: I don't know of a way to do that, but I can say that, as a caller, it is highly annoying. Your agents ought to be able to do that themselves, no? Exactly, otherwise you are losing first chance to make the call different from the other ones where caller feel like they are talking with machines. Simple Hello, it's X, how is your day today sir (and given it's a bit different every day) can change they way the call is going to go... All Asterisk prompts are configurable with a little legwork. Simply use the CLI to see what is playing at the point you want to change, then set up this little ditty to override it. Say you wanted to record the canned tt-weasels prompt (Weasels have eaten our phone system). This 3-liner lets you re-record it. - exten = 999,1,answer - exten = 999,n,record(tt-weasels.gsm) - exten = 999,n,hangup If you are using a codec other than gsm you would replace gsm with wav, slin, etc. -- _ -- 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] ReceiveFax
Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 01/19/2011 02:30 PM, Bryant Zimmerman 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. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM 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). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] ReceiveFax
On 01/20/2011 09:00 AM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Of course ReceiveFAX can be run on multiple channels at once. What makes you think it cannot? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] context problem
Jonas Kellens wrote: [snip] register = 119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten = _52525252,1,NoOp(context TRUNKin - 52525252) exten = _52525252,n,GoTo(blabla,52525252,1) exten = _59595959,1,NoOp(context TRUNKin - 59595959) exten = _59595959,n,GoTo(blablabla,59595959,1) Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? Could you try removing the leading '_', as you seem to be expecting the exact number? Try that and let us know. Regards, -- 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] OT - TTS in spanish
Hi, For an organization welcoming turists (in France), I would be curious to learn about successful use (with Asterisk) of Text-To-Speech in spanish (and english). I took a look at Cepstral's web site and saw there 2 Americas Spanish voices (along a bunch of english voices). 1. In this context, according to your experience, is it acceptable to use an Americas Spanish voice ? 2. Which TTS would you recommend ? 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] ReceiveFax
Hi, I set up ReceiveFax to answer a specific number (2134-4805) , so , the first caller get the fax signal and transmit the fax normal, but, if another caller to call the same number almost at the same time, it gets the signal as well but the fax is not sent! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 20 Jan 2011 09:13:44 -0600 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ReceiveFax On 01/20/2011 09:00 AM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Of course ReceiveFAX can be run on multiple channels at once. What makes you think it cannot? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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
Re: [asterisk-users] context problem
On 01/20/2011 04:43 PM, Jose P. Espinal wrote: Jonas Kellens wrote: [snip] register = 119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten = _52525252,1,NoOp(context TRUNKin - 52525252) exten = _52525252,n,GoTo(blabla,52525252,1) exten = _59595959,1,NoOp(context TRUNKin - 59595959) exten = _59595959,n,GoTo(blablabla,59595959,1) Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? Could you try removing the leading '_', as you seem to be expecting the exact number? Try that and let us know. Regards, Hello, I have tried that yet. It did not make any difference... 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] Hi, agent intro-speech for outside caller
On Thu, 20 Jan 2011, Danny Nicholas wrote: All Asterisk prompts are configurable with a little legwork. Simply use the CLI to see what is playing at the point you want to change, then set up this little ditty to override it. Say you wanted to record the “canned” tt-weasels prompt (“Weasels have eaten our phone system”). This 3-liner lets you re-record it. - exten = 999,1,answer - exten = 999,n,record(tt-weasels.gsm) - exten = 999,n,hangup If you are using a codec other than gsm you would replace gsm with wav, slin, etc. Another technique is to fiddle with the LANGUAGE channel variable. For example (off the top of my head): exten = s,n,set(CHANNEL(language)=mike) exten = s,n,playback(agent-intro) ... Then, record /var/lib/asterisk/sounds/agent-intro.wav to be something generic like Hi. I don't know what my name is, but your call is exceedingly valuable to us and /var/lib/asterisk/sounds/mike/agent-intro.wav to be something more specific like Hi. My name is Mike and I specialize in sounding sincere even when I really don't care. I am extremely sorry you have to wait these brief moments until I am able to provide you with exceptional service today or tomorrow depending on the length of our call queue, you know? -- 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] OT - TTS in spanish
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, January 20, 2011 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT - TTS in spanish Hi, For an organization welcoming turists (in France), I would be curious to learn about successful use (with Asterisk) of Text-To-Speech in spanish (and english). I took a look at Cepstral's web site and saw there 2 Americas Spanish voices (along a bunch of english voices). 1. In this context, according to your experience, is it acceptable to use an Americas Spanish voice ? 2. Which TTS would you recommend ? Regards Just my .02 - unless you need a lot of free form responses, you would probably be better off with a set of pre-recorded prompts. Cepstral actually has more than 2 Spanish voices if I recall correctly, but the end result will sound canned because of the TTS engine. For example, if you take the standard Asterisk prompts recorded by Allison Smith and record them using Cepstral-Allison the results will be significantly different. The Good news is that the tourists probably won't be too critical of the quality either way. -- _ -- 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] ReceiveFax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, January 20, 2011 9:00 AM
To: Asterisk Asterisk
Subject: [asterisk-users] ReceiveFax
Hi all,
I realize that the application Receivefax can't handle with more than one
fax at the same time. In a environment with a lot of fax, some caller get
the signal but the operation can't be completed.
Is there a way to send busy tone to the second caller?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
My guess is no. A possible work-around would be to set a global variable
to indicate that the line is busy and to play a message and hang-up
immediately or to just hangup. Something like this:
- exten = s,1,answer
- exten = s,n,AGI(checkstat.agi) - reset variable if receivefax
died or hungup
- exten = s,n,Gotoif($[ ${FAXINUSE} = YES]?byebye)
- exten = s,n,Set(GLOBAL(FAXINUSE)=YES)
- exten = s,n,receivefax
- exten = s,n,Set(GLOBAL(FAXINUSE)=NO)
- exten = s,n,hangup
- exten = s,n(byebye),playback(im-busy)
- exten = s,n,hangup
--
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Re: [asterisk-users] context problem
On 01/20/2011 04:29 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, January 20, 2011 9:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] context problem
Hello list,
Asterisk 1.6.16.1
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
mailto:119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
mailto:119909:pas...@sip.prov.org/59595959
[119909]
type=friend
host=sip.prov.org
username=119909
defaultuser=119909
secret=passwd
context=TRUNKin
extensions.conf :
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten = _59595959,1,NoOp(context TRUNKin - 59595959)
exten = _59595959,n,GoTo(blablabla,59595959,1)
Problem :
the call always enters : exten = _52525252
and never : exten = _59595959
Why is that ??
Kind regards,
Jonas.
Because this an incoming call. What you are trying to accomplish
should be done via ex-girlfriend logic. The way your dialplan is
set up, it assumes you are dialing 525225252 or 59595959 instead of
receiving a call. Here is how the incoming should read
[TRUNKin]
- exten = s,1,answer
- exten = s/52525252,n,Goto(blabla,52525252,1)
- exten = s/59595959,n,Goto(blabla,59595959,1)
- exten = s,n,verbose(call is not from 5252 or 5959)
Hello,
the following is not working :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s/52525252,n,GoTo(blabla,52525252,1)
exten = s/59595959,n,GoTo(blablabla,59595959,1)
exten = s,n,NoOp(nothing)
CLI shows :
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] --
Executing [s@TRUNKin:1] NoOp(SIP/119909-0688, context TRUNKin -
s) in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] --
Executing [s@TRUNKin:2] NoOp(SIP/119909-0688, 775006 775006)
in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Auto
fallthrough, channel 'SIP/119909-0688' status is 'UNKNOWN'
What else can I try ?
Kind regards,
Jonas.
--
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Re: [asterisk-users] context problem
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} : .*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} : .*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
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Re: [asterisk-users] context problem
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, January 20, 2011 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011 04:29 PM, Danny Nicholas wrote:
_
size=2 width=100% align=center
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, January 20, 2011 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] context problem
Hello list,
Asterisk 1.6.16.1
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[119909]
type=friend
host=sip.prov.org
username=119909
defaultuser=119909
secret=passwd
context=TRUNKin
extensions.conf :
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten = _59595959,1,NoOp(context TRUNKin - 59595959)
exten = _59595959,n,GoTo(blablabla,59595959,1)
Problem :
the call always enters : exten = _52525252
and never : exten = _59595959
Why is that ??
Kind regards,
Jonas.
Because this an incoming call. What you are trying to accomplish should be
done via ex-girlfriend logic. The way your dialplan is set up, it assumes
you are dialing 525225252 or 59595959 instead of receiving a call. Here
is how the incoming should read
[TRUNKin]
exten = s,1,answer
exten = s/52525252,n,Goto(blabla,52525252,1)
exten = s/59595959,n,Goto(blabla,59595959,1)
exten = s,n,verbose(call is not from 5252 or 5959)
Hello,
the following is not working :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s/52525252,n,GoTo(blabla,52525252,1)
exten = s/59595959,n,GoTo(blablabla,59595959,1)
exten = s,n,NoOp(nothing)
CLI shows :
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Executing
[s@TRUNKin:1] NoOp(SIP/119909-0688, context TRUNKin - s) in new
stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Executing
[s@TRUNKin:2] NoOp(SIP/119909-0688, 775006 775006) in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Auto
fallthrough, channel 'SIP/119909-0688' status is 'UNKNOWN'
What else can I try ?
Kind regards,
Jonas.
The call is coming through with the ID 119909 from both trunks. You need to
be able to register the trunks as 119909 and some other number (119910?) or
otherwise you will have to query the SIP headers to get the actual
information from the duplicated trunks (maybe an AGI?)
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Re: [asterisk-users] ReceiveFax
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time.
[fax-in]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten =
s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten = s,n,Set(${LOCALSTATIONID})
exten = s,n,MixMonitor(/mnt/ramdisk/${BASEFILE}.wav)
exten = s,n,ReceiveFAX(/mnt/ramdisk/${BASEFILE}.tif)
exten = s,n,Hangup()
exten = h,1,System(/home/asterisk/dofax.sh ${EMAILADDRESS} ${FAXSTATUS}
${CALLERID(num)} ${snip
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, January 20, 2011 10:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ReceiveFax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, January 20, 2011 9:00 AM
To: Asterisk Asterisk
Subject: [asterisk-users] ReceiveFax
Hi all,
I realize that the application Receivefax can't handle with more than one
fax at the same time. In a environment with a lot of fax, some caller get
the signal but the operation can't be completed.
Is there a way to send busy tone to the second caller?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
My guess is no. A possible work-around would be to set a global variable
to indicate that the line is busy and to play a message and hang-up
immediately or to just hangup. Something like this:
- exten = s,1,answer
- exten = s,n,AGI(checkstat.agi) - reset variable if receivefax died
or hungup
- exten = s,n,Gotoif($[ ${FAXINUSE} = YES]?byebye)
- exten = s,n,Set(GLOBAL(FAXINUSE)=YES)
- exten = s,n,receivefax
- exten = s,n,Set(GLOBAL(FAXINUSE)=NO)
- exten = s,n,hangup
- exten = s,n(byebye),playback(im-busy)
- exten = s,n,hangup
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[asterisk-users] Accessing a 'user' variable via. dialplan.
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of
the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer 201' - but that gives
everything and parsing that isn't really an option.
Anyone know how to access 'variables' (and maybe the contents) directly?
Thanks
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
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Registered in England. No. 27459085.
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Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Accessing a 'user' variable via. dialplan.
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status
of
the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer 201' - but that gives
everything and parsing that isn't really an option.
Anyone know how to access 'variables' (and maybe the contents)
directly?
Thanks
Posted by Joshua Colp dated 12/19/2010, with the subject of Specifying DID
for outbound calls
I'm surprised nobody has suggested using the setvar functionality. It's
extremely useful for stuff like this and would allow you to keep all
CallerID information with the actual configuration of the device.
Using a configuration entry for sip.conf in another response as an example:
[101]
type=friend
username=101
secret=
mailbox=101
callerid=User One 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
setvar=EXTERNAL_CALLERID=User One 3012323434
And then in extensions.conf:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@vitel-outbound)
Of course you could add some sanity checking there to make sure that
${EXTERNAL_CALLERID} contains a value and if not default to your main DID.
-
I think you can get an idea on how to access setvar much easier, he also
stated you can have multiple setvar(s)
Ie,
Setvar=VAR_1=Taco
Setvar=VAR_2=Apples
Setvar=VAR_3=Bannanna
--
William Stillwell
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Re: [asterisk-users] context problem
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} : .*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} : .*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
Hello,
this is the result when using your config :
[Jan 20 17:33:50] -- Executing [s@TRUNKin:1]
NoOp(SIP/119909-06d7, context TRUNKin - s) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:2]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:3]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:4]
NoOp(SIP/119909-06d7, sip:s@11.11.12.112) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:5]
NoOp(SIP/119909-06d7, ) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:6]
NoOp(SIP/119909-06d7, 775006) in new stack
dialplan :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${SIP_HEADER(TO)})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID(num)})
Kind regards,
Jonas.
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Re: [asterisk-users] ReceiveFax
On 01/20/2011 11:00 PM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Receivefax can handle hundreds of calls at one time, if your machine's resources are up to it? Why would there be a restriction of one call? Steve -- _ -- 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] ReceiveFax
From: William Stillwell will...@stillwellsoft.com
Sent: Thursday, January 20, 2011 11:26 AM
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time. [fax-in] exten = s,1,Answer() exten = s,n,Wait(1) exten
= s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten = s,n,Set(${LOCALSTATIONID}) exten =
s,n,MixMonitor(/mnt/ramdisk/${BASEFILE}.wav) exten =
s,n,ReceiveFAX(/mnt/ramdisk/${BASEFILE}.tif) exten = s,n,Hangup() exten =
h,1,System(/home/asterisk/dofax.sh ${EMAILADDRESS} ${FAXSTATUS}
${CALLERID(num)} ${snip From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, January 20, 2011 10:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ReceiveFax From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, January 20, 2011 9:00 AM
Hi all,I realize that the application Receivefax can't handle with
more than one fax at the same time. In a environment with a lot of fax, some
caller get the signal but the operation can't be completed.Is there a way
to send busy tone to the second caller?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda My guess is no. A possible work-around would be to
set a global variable to indicate that the line is busy and to play a message
and hang-up immediately or to just hangup. Something like this: - exten
= s,1,answer - exten = s,n,AGI(checkstat.agi) - reset variable if
receivefax died or hungup - exten = s,n,Gotoif($[ ${FAXINUSE} =
YES]?byebye) - exten = s,n,Set(GLOBAL(FAXINUSE)=YES) - exten =
s,n,receivefax - exten = s,n,Set(GLOBAL(FAXINUSE)=NO) - exten =
s,n,hangup - exten = s,n(byebye),playback(im-busy) - exten =
s,n,hangup
Why can't receivefax handle more then 5 faxes at the same time? Are you using
the res_fax_spandsp.so or the res_fax_digium.so modules? It was my
understanding that the res_fax_spandsp.so did not have a limit and the
res_fax_digium.so was the commercial offering that is based on a per channel
license.
Am I wrong on the res_fax_spandsp.so module is there a limit other than
hardware performance?
Bryant
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Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
That's what I am already using :)
Somehow, the outbound ID sometimes gets messed up (maybe to do with 2
calls from different users at once) - and the wrong one is sent to the
telco.
So, rather than just using a 'Set(CALLERID(num)=callidnum' just before
Dial - I wanted to check the user directly (to double-check Asterisk if
you like and check my own sanity).
Something alone the lines of 'Set(idvar=${SIPPEER(201:callidnum)})' or
even 'Set(idvar=${SIPPEER(201:variables)})' [to parse that little bit
myself]. That way I can check if there is a genuine problem - or if,
indeed, it is the telco themselves (I don't want to leave a Trend tester
on-site).
Thanks anyway.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: 20 January 2011 16:31
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Accessing a 'user' variable via. dialplan.
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status
of the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer 201' - but that
gives everything and parsing that isn't really an option.
Anyone know how to access 'variables' (and maybe the contents)
directly?
Thanks
Posted by Joshua Colp dated 12/19/2010, with the subject of Specifying
DID for outbound calls
I'm surprised nobody has suggested using the setvar functionality. It's
extremely useful for stuff like this and would allow you to keep all
CallerID information with the actual configuration of the device.
Using a configuration entry for sip.conf in another response as an
example:
[101]
type=friend
username=101
secret=
mailbox=101
callerid=User One 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
setvar=EXTERNAL_CALLERID=User One 3012323434
And then in extensions.conf:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@vitel-outbound)
Of course you could add some sanity checking there to make sure that
${EXTERNAL_CALLERID} contains a value and if not default to your main
DID.
-
I think you can get an idea on how to access setvar much easier, he also
stated you can have multiple setvar(s)
Ie,
Setvar=VAR_1=Taco
Setvar=VAR_2=Apples
Setvar=VAR_3=Bannanna
--
William Stillwell
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If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] res_fax
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman 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. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM 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). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve -- _ -- 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] context problem
I always thought the last bit (after the /) is where the context in
sip.conf landed.
What about:
(sip.conf)
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[52525252]
...
context = TRUNKin52
...
[59595959]
...
context = TRUNKin59
...
And split them out in extensions.conf?
I have a suspicion that you have 'context=TRUNKin' under the '[default]'
section of sip.conf - which is why they are hitting there in the first
place.
Then again, I have been known to be wrong ;)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 20 January 2011 16:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} :
.*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} :
.*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
Hello,
this is the result when using your config :
[Jan 20 17:33:50] -- Executing [s@TRUNKin:1]
NoOp(SIP/119909-06d7, context TRUNKin - s) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:2]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:3]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:4]
NoOp(SIP/119909-06d7, sip:s@11.11.12.112) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:5]
NoOp(SIP/119909-06d7, ) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:6]
NoOp(SIP/119909-06d7, 775006) in new stack
dialplan :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${SIP_HEADER(TO)})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID(num)})
Kind regards,
Jonas.
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you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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[asterisk-users] Mailing list question
Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Mailing list question
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list question Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Putting the -- in front of it might make it go away. -- _ -- 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] Mailing list question
On 01/20/2011 12:01 PM, Andrew Thomas wrote: why not just subscribe with an account that doesn't do that like gmail or yahoo ? Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something likedisclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Mailing list question
Let's see :) -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mailing list question -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list question Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Putting the -- in front of it might make it go away. -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Mailing list question
That's my last option Jon. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: 20 January 2011 16:59 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mailing list question On 01/20/2011 12:01 PM, Andrew Thomas wrote: why not just subscribe with an account that doesn't do that like gmail or yahoo ? Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something likedisclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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
Re: [asterisk-users] context problem
On 01/20/2011 10:58 AM, Jonas Kellens wrote: [snip] I have the following registrations : register = 119909:pas...@sip.prov.org/52525252 mailto:119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 mailto:119909:pas...@sip.prov.org/59595959 [snip] Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? I may be wrong here, but I think you can only register once. The last registration received will overwrite the first one. You will need to specify a second entry and register that one separately. This is the same reason you cannot register two devices to the same extension. Have you checked the logs and verified that the SIP provider actually sends 59595959 when you dial that number? Or do you get sent 52525252 no matter what? Someone please correct me if I am wrong here. 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] Mailing list question 2
Sorry about this - testing this disclaimer problem :) -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Mailing list question
Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? It looks like there were underscores on the same line as the -- I think the actual idea is to include '-- ' with nothing else on that line --Don -- Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 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
Re: [asterisk-users] Mailing list question 2
On 20 Jan 2011, at 17:13, Andrew Thomas wrote: Sorry about this - testing this disclaimer problem :) I can give you a POP3 account on my server if it stops you spamming the list?.. S -- _ -- 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] Mailing list question
On Thu, Jan 20, 2011 at 12:03 PM, Danny Nicholas da...@debsinc.com wrote: Putting the -- in front of it might make it go away. If I am not mistaken it should be exactly two dashes followed by a space on a line alone to indicate the end of the mail content. But not all mail readers will honor it. -- -Bob Beers -- _ -- 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] Mailing list question
Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mailing list question -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list question Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Putting the -- in front of it might make it go away. -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Mailing list question 2
Tell you what Steve - I'll not take you up on your kind offer - I'll just let my server keep adding the disclaimer. There - problem solved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 20 January 2011 17:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mailing list question 2 On 20 Jan 2011, at 17:13, Andrew Thomas wrote: Sorry about this - testing this disclaimer problem :) I can give you a POP3 account on my server if it stops you spamming the list?.. S -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Internode weirdness
On 01/19/2011 10:34 PM, Da Rock wrote: 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. Have you tried disallowing re-invites? -- _ -- 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] iNum at 12 Noon EST Friday
Hi, Tomorrow, our discussion is around iNum with lots of interesting people chiming in, including the Voxbone people who manage the space. If you ever wondered about iNum and why you might care about it, how it works, who offers it and who actually uses it, here's a chance to find out more. Join us live at 12 Noon EST or find the local time for you at http://vuc.me/next Call: sip:200...@login.zipdx.com or Skype:vuc.me IRC channel: #vuc on Freenode.net or http://vuc.me/irc for a web client Of course there are iNum numbers: This one from Tropo points you to the right conference: +883510001826724 Join us and contribute your knowledge and experience or learn from others. The VUC guarantee: There is never any top posting on our conferences! /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
[asterisk-users] Polycom 500 / MWI
All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: reginfo msg msg.bypassInstantMessage=1 mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe= /mwi /reginfo and the indicator will come on if there is a new message but it won't go off when I delete the message. I think that after a period of hours it may go off. But the only way to make it go off quickly is to put some invalid chars into the subscribe string and reboot the phone and then switch it back and reboot again. Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? Thanks, Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny
Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init script: #snip-# # Mon Jun 04 2007 Iñaki Baz Castillo i...@in.ilimit.es # - Eliminated SAFE_ASTERISK since it doesn't work as LSB script (it could require a independent safe_asterisk init script). # If you DON'T want Asterisk to start up with terminal colors, comment # this out. COLOR=yes #snop# Commenting out COLOR=yes has no effect. The work around is to use the * 1.4 init script which does call safe_asterisk daemon and things seem to work as expected with the colors. So my question is, will this impact the stability of the system in reference to debian lenny using LSB scripts vs the older init scripts? Or is there another work around to get ssh console colors using the Debian * 1.6.0.28 LSB init script? I also tried 'nocolor = no' in the [options] section of asterisk.conf with no effect. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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 1.6 SSH Console Colors Debian Lenny
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote: Or is there another work around to get ssh console colors using the Debian * 1.6.0.28 LSB init script? I also tried 'nocolor = no' in the [options] section of asterisk.conf with no effect. Try running asterisk using safe_asterisk.. Works for me with 1.4.22 and lenny.. -- _ -- 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] Polycom 500 / MWI
Brian C. Huffman wrote: Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? My phone.cfg for extension 4221 and the voicemail extension of 4200 look like: mwi msg.mwi.1.subscribe=4221@sip msg.mwi.1.callBackMode=registration msg.mwi.1.callBack=4200 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny
On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I'm running Debian but have been running Asterisk since before there was a proper Debian package, and so I ended up writing my own init.d script. See attached. No guarantees or anything :) -- AJS Answers come *after* questions. asterisk Description: application/shellscript -- _ -- 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] Mailing list question
On 01/20/2011 11:16 AM, Andrew Thomas wrote: Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) API provides the physical services and bandwidth for the mailing lists, but does not operate them. If you go to the lists.digium.com site and choose the 'asterisk-users' mailing list, you can see there is a link to send a message to the list administrator(s)... which would probably be more effective than asking a question like this on the list itself :-) In any case, the answer is no... the lists are operated using Mailman software, and it essentially leaves the message bodies alone (although it does do scrubbing of attachments in some cases). Unless you want to include your signature as an attachment marked as something other than 'text', I don't believe there's any way to get the mailing list process to drop your signature block. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Polycom 500 / MWI
On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman bhuff...@etinternational.com wrote: Does anyone know how to setup this phone to work with asterisk so that the indicator light comes on when there's a new message and goes off quickly (less than a minute) after the message is deleted? Thanks, Brian Brian, I'm using Polycom 321 sets, and the MWI works wonderfully. If you look at the asterisk-1.6 source code, in app_voicemail.c, you can see where it calls queue_mwi_event(...) after leaving a message and after deleting a message. If you run a wireshark capture, you should see these in the trace. It also looks like, in most cases, an AMI event of MessageWaiting will be generated. I know it's not much, but it may help you to diagnose the problem further. -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] context problem
I may be wrong here, but I think you can only register once. The last registration received will overwrite the first one. You will need to specify a second entry and register that one separately. This is the same reason you cannot register two devices to the same extension. Yes, that's very likely what is happening. The provider is seeing two SIP registrations arrive, for the same provider account, from the same peer at the same IP address. It is very likely that the second registration is (by design) replacing the first. Then, whenever someone dials a DID associated with this provider account, the provider is routing the call based on the information in the most current registration... it's either going to the context and extension specified in that registration (if their is one) or to the s extension for the relevant context. (Some providers do allow multiple registration for a given account, and will INVITE all of them when an incoming call arrives, but (if I recall correctly) the registrations have to come from different IP addresses (and perhaps different peers) in order to be recognized as being distinct.) There are probably several ways around this: (1) Use two different provider accounts, and associate each DID with a different account. Use two register statements, one per account, and specify different routing extensions on these. (2) Use a provider which will let you register once, and will pass through the DID number which was dialed as the target extension. (3) Use a provider which will let you set up your DIDs for hardwired-IP-address routing (i.e. no register being required) and who passes through the DID as the extension to be called. I recently set up an account with Vitelity, and they support option (3). I simply entered the public IP address of my SIP server for the routing, and everything works correctly... the incoming INVITE requests say sip:MYDID@MYIPADDRESS. Asterisk then uses MYDID as the desired extension in my dialplan, and routes the call appropriately. I'd suggest that the OP ask the current SIP provider whether they handle (2) i.e. whether it's possible for different DIDs associated with a single account to have different information in the INVITE requests sent to the registered client. -- _ -- 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 dos question
I understood that option worked the other way around so attacker thinks peer name is invalid even when they hit a real one. On Wed, Jan 19, 2011 at 2:23 AM, ad...@3a.hu wrote: Hi List, i've been receiving several sip registration probes in the last month, and as this server is a testing site (no external lines, no nothing) i have no fail2ban and still not planning to install. Whenever i have nagios telling me that there is another 'guest', i go and edit iptables manually and that's it. Recently i discovered that these attacks start with some kind of dictionary, and try to guess valid peer names to use one by one. Apparently after quarter million tries, they do find a legitim sip peer name and from that point they stick to that peer name and the attack continues to guess only passwords. Of course, they can not guess passwords like p(F9j43/Qgrhjv*^3 so i'm still not worried, but this made me believe that asterisk responds differently when probing a valid sip peer name. So i was wondering through the sip.conf and found 'alwaysauthreject' which was set to default (commented out). I now set its value to yes (which i thought was the default setting). Does this setting makes the attacker believe that the first try of sip peer name was valid, but only the password was incorrect? So in this case should they stick to the first name tried whatever it was? thanks adam -- _ -- 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 to asterisk t.38
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal -- _ -- 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 to asterisk t.38
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal ami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- 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] Polycom 500 / MWI
I've got the following in my phone.cfg: reginfo msg msg.bypassInstantMessage=1 mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe= /mwi /reginfo The actual config looks good, but the structure of the XML is off. Here's what I use (and it works): phone1 msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=86 / /msg /phone1 You seem to be missing the closing /msg statement, so your XML is not well formed. Also, I don't know what reginfo is, I don't use it in my config. The outermost XML tag for me is phone1, and msg is right inside it. (And of course I left out plenty of other config options from my snippet.) HTH -- _ -- 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 1.8.2.2 Now Available (Security Release)
The Asterisk Development Team has announced a release for the security issue described in AST-2011-001. Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1. This releases is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2 Security advisory AST-2011-001 is available at: http://downloads.asterisk.org/pub/security/AST-2011-001.pdf Thank you for your continued support of Asterisk! -- _ -- 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 dos question
Hi Kyle, On 01-20-2011 20:41, Kyle Kienapfel wrote: I understood that option worked the other way around so attacker thinks peer name is invalid even when they hit a real one. sorry, it must be because i'm not a native english speaker but i don't exactly get what you mean by the above. to me it appears that attackers actually do know when they hit a valid peer name. now i switched the alwaysauthreject to yes (was on default). at the next attack i'll see if they now can determine if a peer name is valid or not. i'm expecting: not from now on. So i was wondering through the sip.conf and found 'alwaysauthreject' which was set to default (commented out). I now set its value to yes (which i thought was the default setting). Does this setting makes the attacker believe that the first try of sip peer name was valid, but only the password was incorrect? So in this case should they stick to the first name tried whatever it was? -- _ -- 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 to asterisk t.38
Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- 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] Introducing easySysAdmin - automated security and telecom fraud protection
Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking of repeated failed attempts. This is done though a local application on your server so your call data is totally secure. In addition, blocked IP addresses are shared via the service so you can pre-emptively block addresses that others experienced attacks from. We offer a FREE one week no obligation trial, and during the month of January we also offer a DISCOUNT for Asterisk users. Just enter voucher asterisk01 at the checkout to receive 30% off. Protection is available from just US$84 per server with the discount. You can read more and give the free trial a go at: http://easysysadmin.com/ Our standard security package includes: - Monitor VoIP traffic (SIP registrations) and block attackers. - Watch remote server access (SSH logins) and block attackers. - Detect spam relay attempts (SMTP) and block attackers. - Scan of network ports to find vulnerabilities. - Check of software for vendor (distribution) security updates. - Custom monitoring of any TCP port or log file you want. - Flexible configuration to set warn and blocking levels. The background to this service is that prior to founding Voisonics I worked with IBM for 10 years, and became responsible for the security planning and audit compliance of many of IBM's voicemail and IVR platforms across the world. Using this experience and knowledge of their standards we have created easySysAdmin. If you have any questions please don't hesitate to contact me directly. Regards, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Ghost ringing
Sorry, this got buried in my inbox. Did you find a fix or the firmware? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 6:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, Is there any chance that you still have this firmware around or can get it? It's unavailable through the Polycom site and through their support. Thanks much in advance, Joe It`s possible the firmware problem is caused by higher (or lower?) latencies. I can only report on my own experience, which is peace and quiet since I switch (I think the good version to have with respect to that issue was 3.3.0) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still experiencing issues; the office isn't. If it were firmware, I'd assume the issue would 'travel' with the phone. Though I can give re-flashing it a shot. Thanks, Joe I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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 -- _ -- 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:
Re: [asterisk-users] Asterisk to asterisk t.38
On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) It sounds like you are trying to send a fax directly from AST 1.6 to AST 1.4 via t.38 that never hits the PSTN. Have you tried sending a fax from AST 1.6 out via the PSTN, and then back in via the PSTN to AST 1.4? 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] res_fax
On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman 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. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM 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). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? -- _ -- 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 to asterisk t.38
Yes Tom, I am sending via the PSTN gateway which is audio code in my case. Thank You Amit Nepal On 1/20/2011 3:07 PM, Tom Rymes wrote: On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) It sounds like you are trying to send a fax directly from AST 1.6 to AST 1.4 via t.38 that never hits the PSTN. Have you tried sending a fax from AST 1.6 out via the PSTN, and then back in via the PSTN to AST 1.4? 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 -- _ -- 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 to asterisk t.38
On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote: On 1/20/2011 3:07 PM, Tom Rymes wrote: On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) It sounds like you are trying to send a fax directly from AST 1.6 to AST 1.4 via t.38 that never hits the PSTN. Have you tried sending a fax from AST 1.6 out via the PSTN, and then back in via the PSTN to AST 1.4? Yes Tom, I am sending via the PSTN gateway which is audio code in my case. Ok, are both boxes connected to the same audiocodes gateway? Or are they in different locations? Can you try sending via regular POTS lines? 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] Asterisk to asterisk t.38
Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT between the two servers or are they on the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all of the time. Our audio codes gateway dumps into a 1.4 box and all faxes calls are then sent to either 1.6.x or 1.8.x boxes and then on to the final ata. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Amit Nepal ami...@phoenixinternet.net Sent: Thursday, January 20, 2011 4:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk to asterisk t.38 Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- 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
Re: [asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection
On 01/20/2011 03:29 PM, David Cunningham wrote: Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. As the description of this mailing list says, it is for *NON-COMMERCIAL* usage. Please do not post advertisements for commercial products to this list. Thank you. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] res_fax
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman 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. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM 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). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve -- _ -- 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 20, 2011, at 8:53 PM, Steve Underwood On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman 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. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM 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). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve Steve thanks for your response. Do I need a copy of spandsp installed or is the res_fax_spandsp.so the complete package. If I need spandsp what version should I be using? The version I compiled and am using is now over a year old spandsp-0.0.5pre4. Where can I get the current stable version with a list of dependencies for compilation? Thanks Bryant -- _ -- 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 client floods port 5060 and gets blocked
On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens jonas.kell...@telenet.bewrote: On 10/28/2010 12:52 PM, Gordon Henderson wrote: On Thu, 28 Oct 2010, Jonas Kellens wrote On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Is it s Snom phone? I've seen Snoms do this... Gordon I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28... Yes, I have seen this with Snom 370s... It's maddening. I'm going to start testing out the version 8.x firmware. - Julian -- _ -- 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
