Re: [asterisk-users] AGI perl script set timeout within script?
Net::DNS::Async is a fire-and-forget asynchronous DNS helper. That is, the user application adds DNS questions to the helper, and the callback will be called at some point in the future without further intervention from the user application. The application need not handle selects, timeouts, waiting for a response or any other such issues. If the same query is added to the queue more than once, the module may combine the queries; that is, it will perform the query only once, and will call each callback registered for that query in turn, passing the same Net::DNS::Response object to each query. For this reason, you should not modify the Net::DNS::Response object in any way lest you break things horribly for a subsequent callback. This module is similar in principle to POE::Component::Client::DNS, but does not require POE. I think you'll like the part about The application need not handle selects, timeouts... :-) /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
8 jan 2010 kl. 08.01 skrev Tilghman Lesher:
On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the
Asterisk dialplan. ?I would like a timeout of 1 second, then return.
Ah, but Perl isn't actually doing the DNS lookup. If you examine his
script, he's merely passing back a name to the Asterisk process, which
is then calling inet_aton(), which is the reason why he cannot control
it from within the script. What he'd actually need to do is to start
using Net::DNS to do the resolution on that name, first, perhaps even
going as far as to connect to the server himself, and relay the channel
between the AGI interface and the remote TCP interface.
Then, he could use alarm() or the Time::Hires module to ensure his own
timeouts override the builtins. But as it stands now, it's all
Asterisk.
I tried the perl script eval, alarm,
$SIG{ALRM} functions till I was blue in the face from cussing at the
screen. It does not appear that the perl script is doing the DNS query,
otherwise the eval alarm would timeout and pass control back to asterisk.
Another indication is that '#define MAX_AGI_CONNECT 2000' in res_agi is not
being invoked because the timeout is around 30 seconds. Is that 30 second
timeout built into Asterisk? Can I put an absolute timeout on an agi
script from the dialplan prior to calling the agi application? Maybe I'll
fork a macro with a timeout, yea, that's it, let start forking, something
new to cuss at.
No, the timeout is built into glibc. I don't see any documented method for
altering it, sorry. The only way to really do it in a way where you can
control the timeouts would be to do it in your Perl script, in the way that I
described above.
Earlier in this thread the settings in resolv.conf was mentioned, which is
where you normally configure the local resolver, unless you bypass it somehow.
From the linux man page:
options timeout:n
sets the amount of time the resolver will wait for a response from a remote
name server before retrying the query via a different name server. Measured in
seconds, the default is RES_TIMEOUT (currently 5, see resolv.h).
options attempts:n
sets the number of times the resolver will send a query to its name servers
before giving up and returning an error to the calling application. The default
is RES_DFLRETRY (currently 2, see resolv.h).
So the timeout depends on the numbers of servers in resolv.conf and the number
of attempts per server multiplied with the timeout - if I understand it
correctly.
/O
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[asterisk-users] How to recieve number returned by $AGI-wait_for_digit($timeout)
hi, i use $AGI-wait_for_digit($timeout) to wait for the user press key 1 ,and then to do something. but how can i get the return number ? is that use $key = $AGI-wait_for_digit($timeout) and $key will be 200 result=49 if i pressed number 1? Thanks! -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to recieve number returned by $AGI-wait_for_digit($timeout)
On 8 Jan 2010, at 09:14, Zhang Shukun wrote:
i use $AGI-wait_for_digit($timeout) to wait for the user press key 1
,and then to do something.
but how can i get the return number ?
is that use $key = $AGI-wait_for_digit($timeout)
and $key will be 200 result=49 if i pressed number 1?
$key = $AGI-wait_for_digit($timeout);
if (chr($opt) eq '1') {
# do stuff..
}
W
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Re: [asterisk-users] Please remove me from the mailing list.
On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Different Steve!! Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
This is what I was using at the time: asterisk-1.4.21.2 libpri-1.4.6 wanpipe-3.2.7 zaptel-1.4.11 spandsp 0.0.4pre16 unknown rx_fax version. As I can see there is a 0.0.6pre16 version now.. At the time, I used this tutorial I found on the net to setup rxfax/spandsp: http://www.asteriskguru.com/tutorials/spandsp.html Which in its self, is extremely old. I was using PSTN. I cut the error rate tremendously when I rebuild the box as there was a pci-express timing issue (I had a 50% fail rate before that), as the PSTN hardware provided determine from examining server logs. But some callers just could not send a fax, it would fail every time, and I just couldn't reproduce it.. I look forward to retrying out spandsp on my next build or server upgrade. Thank you Steve for responding to my post, and all the work you put into SpanDSP the Asterisk Community. William Stillwell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Underwood Sent: Thursday, January 07, 2010 8:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On 01/08/2010 06:05 AM, William Stillwell (Lists) wrote: Has there been any improvement with app_fax ? I stopped using it as I had a high failure rate with inbound faxes (10%+) 1000 faxes a week ,with over a 100 failures can get quite annoying from people complaining.. I could get it to fail everytime I tried sending a solid black fax page. (ie, take a sheet of paper that is all black, or heavily black, and fax it, I got a ton of errors, or just plain rx reception failure).. If you want an idea of the performance level to expect from app_fax see http://www.soft-switch.org/spandsp-soft-fax-performance.html If 10% of your FAXes are failing, and they really have the potential to succeed (i.e. not voice calls, wrong numbers, etc), that's awful. Anything above 1% is poor. app_fax on a well set up system, with no timing issues, achieves 99% success for PSTN calls. Results with calls on the internet will vary, depending on the quality of your VoIP links. T.38 calls are generally far more reliable than audio ones across the internet. Are you using PSTN or VoIP connexions? Sending a black page is no harder than sending a white one. If you want a real stress test, try the checkerboard pattern TIFF file page, amongst the spandsp test data. That takes about half an hour to send one page. On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. If you are going to ask if something has improved, its rather important to say which versions you are running now, and how you use them. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, January 07, 2010 10:30 PM What about: 1) Fixing the slow responding DNS server? 2) Tweaking /etc/resolv.conf options? 3) Setting up a caching name server on your Asterisk host? I'm out of my element on the rest of the thread but have had great success in solving various DNS woes with Asterisk (and several other apps) by simply installing and configuring dnsmasq as a local resolver talking to my primary DNS servers. If the box is up it can talk to a DNS server (itself) and get a response (nxdomain at worst) to allow the app to move on instead of waiting around... sl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
On 8 Jan 2010, at 13:52, John Novack wrote: Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) Different Steve!! I agree with him though :P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On Fri, Jan 8, 2010 at 8:59 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: This is what I was using at the time: asterisk-1.4.21.2 I really, really prefer the faxing in 1.6. It's so nice to configure compared to 1.4. I'll leave it to the ChangeLog and anybody else who wants to chime in on actual differences. I was using PSTN. Great. Because trying to track down voip faxing problems is much worse. hardware provided determine from examining server logs. But some callers just could not send a fax, it would fail every time, and I just couldn't reproduce it.. Did you ever record your faxes? When I was troubleshooting things, I started recording 100% of faxes, and then just blowing them away after a few days with a cron. If I wanted to go back and troubleshoot a particular customer, I could filter by their calls and listen to what was going on. It was amazing how lousy some of the faxes were and it was obviously the customer's fault. I never would have been able to tell that without listening to the audio recordings of the fax transmission. In other cases, it was robodialers wardialing the world, and they weren't even sending a fax. I discovered I had to be VERY careful how I calculated error rate. If you count by absolute successes and failures, the early failure rate looked awful. This was directly correlated to the customers with crap connections retrying the same faxes that were never going to succeed over and over again. When I instead sorted successes and failures by sending phone number, I got very high 90s success rate. This of course, also requires that you're keeping logging in a way that makes this kind of diagnosis possible. Hopefully you have good records. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cheap femtocell's ahead
http://www.pcworld.com/article/186308/magicjack_harnesses_femtocell_for_voip.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
About what? On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote: On 8 Jan 2010, at 13:52, John Novack wrote: Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) Different Steve!! I agree with him though :P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [VUC] Today at 12 Noon EST (6PM CEST, 9AM PST) iNum with Voxbone
Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say should because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be Hacking VoIP Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on IRC #vuc on Freenode.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
On 8 Jan 2010, at 16:03, Randy R wrote: On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote: On 8 Jan 2010, at 13:52, John Novack wrote: Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) Different Steve!! I agree with him though :P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users About what? :-| About dirty top-posters? W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I would have read your message but I couldn't find it amongst all of this garbage... :) -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne Sent: Friday, January 08, 2010 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please remove me from the mailing list. On 8 Jan 2010, at 16:03, Randy R wrote: On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote: On 8 Jan 2010, at 13:52, John Novack wrote: Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) Different Steve!! I agree with him though :P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users About what? :-| About dirty top-posters? W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.
Hello everyone, I'm trying to turn up a SIP trunk with a Cisco UCM (Unified Communications Manager/Call Manager). It's currently configured for 3rd party call control (3pcc). The INVITEs show up without an SDP... Neither the Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified Communications Manager 7.0? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
On Thu, 7 Jan 2010, David Gibbons wrote: Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn't make me scroll through 10 thousand messages and awful rsa keys to get to the message... FLAME AWAY!!! This is not intended as a flame... I just got a gmail account a month ago, and haven't used it but for a single google group and calendar notifications. This morning, after seeing the above message, I actually hit reply on several messages, and this is what I found: 1) In every case, gmail presented me with the entire text of the message in the compose window. There was NO indication of 'hidden' full-quote. Yes, the cursor is initially placed at the top of the window. 2) The 'Daily Agenda' mails I get from Google Calendar arrive in some kind of rich formatting, but right at the top of the composer window is a small unobtrusive link labelled 'Plain text', which strips the formatting, and makes deleting the unnecessary text trivial. 3) Plain text email arriving from a friend's android/gmail device are displayed in plain text already. 4) I searched thru the settings dialog, and I found nothing where I had explicitly told it to include the text in a reply, or to show or hide that text. I DID specify that 'plain text' was to be my default outgoing format. IMHO, top-posting isn't the problem, but just an obvious symptom of the real problem, which is failure to edit/strip the quotes to the bare minimum. When a thread gets hijacked by top-posters, who bang out their thoughts without even scrolling down to see all the garbage below, another problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! -- Rick Green Those who would give up essential Liberty, to purchase a little temporary Safety, deserve neither Liberty nor Safety. -Benjamin Franklin As for our common defense, we reject as false the choice between our safety and our ideals. -President Barack Obama 20 Jan 2009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I have been doing this (whatever that is), since about 1976, involving many facets, including posting on #1 CBBS out of Illinois, usenet in the 90s, and more. It is not possible to get people to follow all the RFC rules and customs much less the -- CR sigdashes. There are a lot of relative newbies much less oldies who never heard of such, or run 10 different mail options from gmail to hotmail to sendmail to I have no idea what it is, I am just a member, poster, customer, or-something mail user. In the 90s, a very well like member of a BBS type system (MajorBBS/Worldgroup) went ballistic when people started using HTML. The other people on the net finally just told him We don't care, we are not staying in the dark ages. Like it or lump it.. I am on numerous lists where 75% to 100% of the posts are top posts. If someone bottom posts people who want to go to the bottom and read. Most of us start at the earliest post and read message by message.. and don't want to rescroll through 10-20 messages over and over. I know that it would be nice to have the last message have all the text inline, but that doesn't happen either. And then there are always 5 more messages in the same thread later today. I don't even use a sig file. I just type my name. But to see if it works: -- Cary Fitch IMHO, top-posting isn't the problem, but just an obvious symptom of the real problem, which is failure to edit/strip the quotes to the bare minimum. When a thread gets hijacked by top-posters, who bang out their thoughts without even scrolling down to see all the garbage below, another problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! -- Rick Green Those who would give up essential Liberty, to purchase a little temporary Safety, deserve neither Liberty nor Safety. -Benjamin Franklin As for our common defense, we reject as false the choice between our safety and our ideals. -President Barack Obama 20 Jan 2009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Rick Green wrote: On Thu, 7 Jan 2010, David Gibbons wrote: Yes, gmail DOES default to top posting, because bottom posting is silly (in general, but especially for a client that hides quoted text (like gmail)). Top posting is modern. And better. And doesn't make me scroll through 10 thousand messages and awful rsa keys to get to the message... FLAME AWAY!!! This is not intended as a flame... I just got a gmail account a month ago, and haven't used it but for a single google group and calendar notifications. This morning, after seeing the above message, I actually hit reply on several messages, and this is what I found: 1) In every case, gmail presented me with the entire text of the message in the compose window. There was NO indication of 'hidden' full-quote. Yes, the cursor is initially placed at the top of the window. 2) The 'Daily Agenda' mails I get from Google Calendar arrive in some kind of rich formatting, but right at the top of the composer window is a small unobtrusive link labelled 'Plain text', which strips the formatting, and makes deleting the unnecessary text trivial. 3) Plain text email arriving from a friend's android/gmail device are displayed in plain text already. 4) I searched thru the settings dialog, and I found nothing where I had explicitly told it to include the text in a reply, or to show or hide that text. I DID specify that 'plain text' was to be my default outgoing format. IMHO, top-posting isn't the problem, but just an obvious symptom of the real problem, which is failure to edit/strip the quotes to the bare minimum. When a thread gets hijacked by top-posters, who bang out their thoughts without even scrolling down to see all the garbage below, another problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! And to add on to this: aside from whether you think it is silly or not, there are: 1) RFC's 2) List rules And when both of those tell you to bottom-post, then who are you to decide otherwise, just because you think it is silly? Well, maybe I think it is silly that I cannot hit you in the face everytime you say I, would you allow me to hit you, or would you protest and demand I keep to the rules that tell me I can't do that? Civility demands I keep to the rules and do not hit you in the face. The same civility demands you keep to the rules as well and do not top-post! Is that *really* so hard? Just because Microsoft and others decide to place the cursor at the wrong position doesn't mean you have to be a mindless herd-animal and follow that incorrect behavior! Please people, stop these totally pointless discussions and get back on-topic!... PS: I did not have to cut anything, thanks to Rick using the dash-dash-space convention, and Thunderbird honoring that convention. PPS: Top or Bottom posting does NOT change anything about the fact you should SNIP stuff that is no longer relevant Just my €0.02! -- Francesco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
And how will we ever re-write the 10+-year-old RFCs which no longer hold relevance to modern email clients if nobody goes against the grain and does what makes sense rather than what has been generally accepted? -Dave snip And to add on to this: aside from whether you think it is silly or not, there are: 1) RFC's 2) List rules And when both of those tell you to bottom-post, then who are you to decide otherwise, just because you think it is silly? Well, maybe I think it is silly that I cannot hit you in the face everytime you say I, would you allow me to hit you, or would you protest and demand I keep to the rules that tell me I can't do that? /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons d...@videon-central.com wrote: I would have read your message but I couldn't find it amongst all of this garbage... Funny I saw your right away :) Ok, all kidding aside, I really don't care where people post if only they'd clip all the garbage out, the footers, the greetings, etc and just left the points to answer and their answers. But heck, I'm being serious which is against all the RFC. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free FaxForAsterisk ReceiveFAX not working
Hello srinivas,
I have the same issue on my Asterisk installation (Asterisk 1.4.25).
Today i've updated both res_fax.so and res_fax_digium to latest version but
no success.
pfunitasbh*CLI fax show version
FAX For Asterisk Components:
Applications: 1.4_1.1.6
Digium FAX Driver: 1.4_1.1.6 (optimized for pentium_m_32)
Did you checked in Asterisk Console if the T38 module of free fax is
enabled?
fax show capabilities
My Output:
--
Registered FAX Technology Modules:
Type: DIGIUM
Description : Digium FAX Driver
Capabilities: SEND RECEIVE G.711
--
On another installation (output found on google) the response of command is:
--
Registered Fax Technology Modules:
Type : T.38
Description : Digium Fax T.38 Driver
Capabilities : SEND, RECEIVE, UDP
Type : G.711
Description : Digium Fax G.711 Driver
Capabilities : SEND, RECEIVE
--
As you can see, the T38 module isn't enabled on my installation. Tried ask
google how to make it work, but found no hints yet.
Anyone can help us?
Thanks!
Daniel Araujo
May the Source be With You!
Linux user #433396
http://counter.li.org
2010/1/4 srinivas Antarvedi srinivas.antarv...@gmail.com
Hello users,
Recently i have installed the free version of FaxForAsterisk and trying to
work with it by sending a fax
on T38.
My version information is as follows
i)Asterisk 1.6.0.20
ii)res_fax-1.6.0.14_1.1.6-x86_32
iii)res_fax_digium-1.6.0.14_1.1.6-i686_32
sip.conf
[general]
t38pt_udptl=yes
extensions.conf
[default]
exten = _XX,1,NoOp(Fax Incoming Call)
exten = _XX,n,GoTo(faxin,${EXTEN},1)
[faxin]
exten = _XX,1,NoOp(This is ReceiveFAX application Testing)
exten = _XX,n,Wait(6)
exten = _XX,n,NoOp(*** SETTING FAXOPTS *)
exten = _XX,n,Set(FAXOPT(ecm)=yes)
exten = _XX,n,Set(FAXOPT(localstationid)= 1234567890)
exten = _XX,n,Set(FAXOPT(maxrate)=14400)
exten = _XX,n,Set(FAXOPT(minrate)=2400)
exten = _XX,n,Set(FAXOPT(modem)=V17)
exten = _XX,n,Wait(6)
exten = _XX,n,NoOp(* RECEIVING FAX *)
exten = _XX,n,ReceiveFAX(/root/receivefax.tif)
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(locastationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid): ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(err) : ${FAXOPT(error)})
and my asterisk cli information is as follows
asterisk CLIfax show version
FAX For Asterisk Components:
Applications: 1.6.0.14_1.1.6
Digium FAX Driver: 1.6.0.14_1.1.6 (optimized for i686_32)
Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [2016289...@default:1] NoOp(SIP/204.16.59.19-0014, Fax
Incoming Call) in new stack
-- Executing [2016289...@default:2] Goto(SIP/204.16.59.19-0014,
faxin,2016289913,1) in new stack
-- Goto (faxin,2016289913,1)
-- Executing [2016289...@faxin:1] NoOp(SIP/204.16.59.19-0014, This
is ReceiveFAX application Testing) in new stack
-- Executing [2016289...@faxin:2] Wait(SIP/204.16.59.19-0014, 6)
in new stack
-- Executing [2016289...@faxin:3] NoOp(SIP/204.16.59.19-0014,
*** SETTING FAXOPTS *) in new stack
-- Executing [2016289...@faxin:4] Set(SIP/204.16.59.19-0014,
FAXOPT(ecm)=yes) in new stack
-- Executing [2016289...@faxin:5] Set(SIP/204.16.59.19-0014,
FAXOPT(localstationid)= 1234567890) in new stack
-- Executing [2016289...@faxin:6] Set(SIP/204.16.59.19-0014,
FAXOPT(maxrate)=14400) in new stack
-- Executing [2016289...@faxin:7] Set(SIP/204.16.59.19-0014,
FAXOPT(minrate)=2400) in new stack
-- Executing [2016289...@faxin:8] Set(SIP/204.16.59.19-0014,
FAXOPT(modem)=V17) in new stack
-- Executing [2016289...@faxin:9] Wait(SIP/204.16.59.19-0014, 6)
in new stack
-- Executing [2016289...@faxin:10] NoOp(SIP/204.16.59.19-0014,
* RECEIVING FAX *) in new stack
-- Executing [2016289...@faxin:11] ReceiveFAX(SIP/204.16.59.19-0014,
/root/receivefax.tif) in new stack
-- Channel 'SIP/204.16.59.19-0014' receiving FAX '/root/receivefax.tif'
[Dec 31 17:39:55] NOTICE[23578]: res_fax.c:712 generic_fax_exec:
Negotiating T.38 for receive on SIP/204.16.59.19-0014
[Dec 31 17:39:55] NOTICE[23578]: res_fax.c:779 generic_fax_exec: Negotiated
Re: [asterisk-users] Free FaxForAsterisk ReceiveFAX not working
On Fri, Jan 8, 2010 at 1:47 PM, Daniel Araujo redsna...@gmail.com wrote: I have the same issue on my Asterisk installation (Asterisk 1.4.25). As you can see, the T38 module isn't enabled on my installation. Tried ask google how to make it work, but found no hints yet. Anyone can help us? If you want T.38, you should be using 1.6. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Rick Green wrote: problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! Good idea, done! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
And some of the most rabid bottom posters are as guilty of snipping out all the mailing list garbage added to every message. Scrolling through 5 of these to find some comment is as, or more annoying that top posting. Randy R wrote: On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons d...@videon-central.com wrote: I would have read your message but I couldn't find it amongst all of this garbage... Funny I saw your right away :) Ok, all kidding aside, I really don't care where people post if only they'd clip all the garbage out, the footers, the greetings, etc and just left the points to answer and their answers. But heck, I'm being serious which is against all the RFC. /r Posting here as well to please the dinos! And some of the most rabid bottom posters are as guilty of snipping out all the mailing list garbage added to every message. Scrolling through 5 of these to find some comment is as, or more annoying that top posting. -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue_log file and mysql logs together!
Hi, I'm trying to registers events of queues in /var/log/asterisk/queue_log and Mysql database .I have configured realtime queue_log on MySQL and works well, but /var/log/asterisk/queue_log file is empty, since you're not registering events of queues. Removing extconfig.conf configurations (queue_log = mysql,general), /var/log/asterisk/queue_log works well, events logs on /var/log/asterisk/queue_log . With extconfig.conf configurations no events logs on /var/log/asterisk/queue_log. What happens?? My asterisk version is 1.6.1.11. addons 1.6.1.2 res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = userX dbpass = passX dbport = 3306 dbsock = /tmp/mysql.sock -- extconfig.conf [settings] queue_log = mysql,general logger.conf [general] queue_log = yes queue_log_name = queue_log Thanks, Best regards!! Cristian Arguello. __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4755 (20100108) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
In version prior to 1.6, timing is very critical for faxing, and the use of a timing source improves fax sending/receiving., and if no timing source was used, then you would use zt_dummy, but I am not sure how reliable that is or was.. And from what I am reading, v1.6 is far better with faxing, and I would assume the res_timing_*.so is an improved version of the later zt_dummy timing source. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan Vrban Sent: Friday, January 08, 2010 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
yeah, but what about internal_timing = yes in asterisk.conf yes or no for faxing? Or is this option irrelevant for app_fax/spandsp ? 2010/1/8 William Stillwell (Lists) william.stillwell-li...@ablebody.net: In version prior to 1.6, timing is very critical for faxing, and the use of a timing source improves fax sending/receiving., and if no timing source was used, then you would use zt_dummy, but I am not sure how reliable that is or was.. And from what I am reading, v1.6 is far better with faxing, and I would assume the res_timing_*.so is an improved version of the later zt_dummy timing source. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan Vrban Sent: Friday, January 08, 2010 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
See this thread: http://lists.digium.com/pipermail/asterisk-dev/2006-April/019756.html you still need the ztdummy if no hardware timer is not available, and from what I read, internal_timing=yes tells it use the hardware timer if available. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan Vrban Sent: Friday, January 08, 2010 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? yeah, but what about internal_timing = yes in asterisk.conf yes or no for faxing? Or is this option irrelevant for app_fax/spandsp ? 2010/1/8 William Stillwell (Lists) william.stillwell-li...@ablebody.net: In version prior to 1.6, timing is very critical for faxing, and the use of a timing source improves fax sending/receiving., and if no timing source was used, then you would use zt_dummy, but I am not sure how reliable that is or was.. And from what I am reading, v1.6 is far better with faxing, and I would assume the res_timing_*.so is an improved version of the later zt_dummy timing source. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan Vrban Sent: Friday, January 08, 2010 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxing: Anyone have a compiled executable? On a well set up system you should be able to send or receive those pages all day. If you can't, you probably have timing issues in your Asterisk setup. This is a uncleared question. What does timing issue exactly mean? 1) Enable internal timing and use one of the res_timing_*.so (with asterisk =1.6.2) 2) Does it mean, use only one of the res_timing_*.so and no internal timing ? 3) Or something completely different? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multicast RTP Paging
HI Guys, I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast. https://issues.asterisk.org/view.php?id=11797 http://svnview.digium.com/svn/asterisk?revision=101218view=revision My dialplan command is as below. exten = 1234,1,RTPPage(basic|224.1.1.1:7000|ulaw|ef) i have the same IP/Port to be listened on for multicast traffic on the Snom 320's. But when i make a call to 1234, the snom 320 does not get answered at all. If i use the same command and the IP of the Snom instead of the multicase IP, i was able to have the snom auto answer the call on Speaker. I would like get assistance from the community in this issue. Thanks as always Regards Krishna On Wed, May 13, 2009 at 9:21 AM, Joshua Colp jc...@digium.com wrote: Hello everyone, A month ago I took on an issue on the Asterisk issue tracker ( https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP paging. This is the ability to send audio to phones (the phone must support it) and have it played out the speakerphone. Using multicast RTP is great for this because it does not incur the cost and weight of setting up a potentially short call. Depending on the setup this can actually get to be quite a big problem because when you involve phones subscribed to the state of another they get told that the phone is in use. The amount of SIP traffic can just spiral out of control. Originally this issue was filed with a new application that performed the paging. I took this application and turned it into a channel driver. This means that instead of having a dedicated paging application for it you can just use Dial(). This also means that in mixed environments you can use the Page() application along with other phones that do not support the multicast RTP paging. So far I have gotten very little response on the issue so I am asking anyone on this mailing list who is interested and has the time to test to please test and provide some feedback. A branch based off of trunk (as that is where the channel driver will go) is available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797 The dial string for the channel driver is in the form of MulticastRTP/type/destination/control address where type is either basic or linksys. The control address is only needed for the linksys type. Any feedback is welcome as a note on https://issues.asterisk.org/view.php?id=11797 and will help to getting this into the tree. Thanks! -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I get codec info on active calls
Hello All. I would like to know what codec is being used during a call. For example if I have 3 channels on 3 active calls how can I find what codec is beeing used by each client? Thanks in advanced. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
What about: 1) Fixing the slow responding DNS server? 2) Tweaking /etc/resolv.conf options? 3) Setting up a caching name server on your Asterisk host? 4) Adding the AGI server host name and IP address to /etc/hosts? 5) Using the IP address of the AGI server in your dialplan? Ok, I went with #4 for a bit, then resolved to #5 (pardon the pun), works fine. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I get codec info on active calls
On Fri, 8 Jan 2010, Landy Landy wrote: I would like to know what codec is being used during a call. For example if I have 3 channels on 3 active calls how can I find what codec is beeing used by each client? How about something like: asterisk -r -x sip show channel blah-blah-b...@a.b.c.d\ | grep Format -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multicast RTP Paging
Hi! I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? Are you asking us if you are trying to do this? Only you would know. ;-) i have the same IP/Port to be listened on for multicast traffic on the Snom 320's. But when i make a call to 1234, the snom 320 does not get answered at all. If i use the same command and the IP of the Snom instead of the multicase IP, i was able to have the snom auto answer the call on Speaker. Have you first tested the SNOM multi-cast feature with either VLC or MAST to make sure it is set up correctly? Details are here: http://www.voip- info.org/wiki/index.php?page=Asterisk+phone+snom#RelatedMulticastapp_rtppa geAsterisk16orl Note: For the SNOM this is not a phone call, it does therefore not answer; all you get is a remote speaker without local volume control, and without any entiers in the call list/history. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
