Re: [asterisk-users] AGI perl script set timeout within script?

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 th

Re: [asterisk-users] AGI perl script set timeout within script?

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. On Thu, 7 Ja

Re: [asterisk-users] AGI perl script set timeout within script?

> >> 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. > >> > > On Thur

Re: [asterisk-users] Please remove me from the mailing list.

Careful, or Steve will "un top post" YOU! David Gibbons wrote: > > I haven’t had a good mailing list war in a while. > > 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 moder

Re: [asterisk-users] AGI perl script set timeout within script?

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. On Thursday 07 January 2010 18:59:24 D

Re: [asterisk-users] AGI perl script set timeout within script?

On Thursday 07 January 2010 18:59:24 David Backeberg 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 time

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 f

Re: [asterisk-users] voicemail /odbc problem

On Thursday 07 January 2010 16:20:38 Alex Sharaz wrote: > I'm having a bit of a problem with storing voicemail messages in an > odbc database. I *think* I've got everything configured correctly but > messages are stored on the asterisk server instread of in the database. > > System info > > 64 bit

Re: [asterisk-users] AGI perl script set timeout within script?

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. A few things... * stop using DNS? Pro

Re: [asterisk-users] Realtime LDAP Queues crashes

What are the LDAP searches like? On 05/01/2010, Jorge Salamero Sanz wrote: > Hi all, > > I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other > attributes needed for a working LDAP backend (I'll open a bug to include > these > changes on svn). > > SIP users and dialplan are

Re: [asterisk-users] AGI perl script set timeout within script?

On Thu, 7 Jan 2010, JR Richardson wrote: > I'm running an AGI, calling a perl script the does number lookups to a > remote server. I would like to put a timeout in the script. The > problem I'm running into is if the DNS server is not responding, the > script hangs and waits for 30 seconds be

[asterisk-users] AGI perl script set timeout within script?

Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The 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

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

On Thu, Jan 07, 2010 at 05:05:09PM -0500, William Stillwell (Lists) wrote: > Has there been any improvement with app_fax ? > Builds with spandsp 0.0.6 (as opposed to older versions that required older versions of spandsp). It is also a more well-behaving Asterisk app in its logging (does not kee

[asterisk-users] voicemail /odbc problem

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed use

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

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

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: > At 00:22 1/7/2010, Tzafrir Cohen wrote: > >On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: > >> At 16:49 1/5/2010, Tzafrir Cohen wrote: > >> >On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote: > >> >> Hi, > >> >> > >> >> Hav

Re: [asterisk-users] Crash in Asterisk

On Thu, 7 Jan 2010 15:58:43 -0430 Danny Dias wrote: > My friends, > > I'm having some problems in my Asterisk, the thing is that Asterisk > seem to be crashed (or dead) sometimes (2 times in 3 weeks) > > [Jan 5 16:51:19] WARNING[6787] channel.c: Channel allocation failed: > Refusing due to ac

[asterisk-users] Crash in Asterisk

My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in co

Re: [asterisk-users] Please remove me from the mailing list.

http://www.washington.edu/computing/mailman/faqs/mailman.email.html Em 07/01/2010, às 15:29, "C. Chad Wallace" escreveu: > > At 2:01 PM on 07 Jan 2010, Dan Journo wrote: > >> I've never seen that in Outlook. What client do you use? > > Claws Mail provides a "Mailing-List" sub-menu under the M

Re: [asterisk-users] Please remove me from the mailing list.

I haven't had a good mailing list war in a while. 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

Re: [asterisk-users] Please remove me from the mailing list.

On 7 Jan 2010, at 19:01, Dan Journo wrote: > I've never seen that in Outlook. What client do you use? He said 'proper' mail client ;) *holy war* Sorry... S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mail

Re: [asterisk-users] Please remove me from the mailing list.

On Thu, Jan 7, 2010 at 1:27 PM, Warren Selby wrote: > I use gmail but don't see any buttons for unsubscribe or anything like that? Click on 'show details' at the top of the message and it will expand to show those options. I just found them over Christmas as I was trying to thin down the number

Re: [asterisk-users] Please remove me from the mailing list.

At 2:01 PM on 07 Jan 2010, Dan Journo wrote: > I've never seen that in Outlook. What client do you use? Claws Mail provides a "Mailing-List" sub-menu under the Message menu, which includes Post, Subscribe and Unsubscribe options, among others. It's amazing what paying attention to standards can

Re: [asterisk-users] Please remove me from the mailing list.

I use gmail but don't see any buttons for unsubscribe or anything like that? Also, gmail defaults to top posting...which seems to upset some people 'round these parts. I have yet to find a way to make gmail not top-post by default... On Thu, Jan 7, 2010 at 1:16 PM, Francesco Peeters wrote: > D

Re: [asterisk-users] Please remove me from the mailing list.

Dan Journo wrote: > I've never seen that in Outlook. What client do you use? > > Lately I have been using Thunderbird with an RFC2369 header plugin. --FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailin

Re: [asterisk-users] Please remove me from the mailing list.

Gmail DOES process those headers... > >And a proper mail client will also parse the headers and provide unsubscribe >information/buttons based on that ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing li

Re: [asterisk-users] Please remove me from the mailing list.

I've never seen that in Outlook. What client do you use? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco Peeters Sent: 07 January 2010 18:58 To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Please remove me from the mailing list.

Steve Totaro wrote: > read your posting and it will tell you haw to remove yourself. > > On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean > wrote: > > Can I be taken off the mailing list please. > > Thanks. > rick > > http://lists.digium.com/mailman/listinfo/asteri

[asterisk-users] dns messages on console

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup

Re: [asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone

Ok, im gonna go craw back under a rock.. Third line of my sip.conf allowtransfer=no Thanks for those who responded (Steve & Ollie) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E. Johansson Sent: Th

Re: [asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone

7 jan 2010 kl. 17.15 skrev William Stillwell (Lists): > I have several sip stations that on a that are on a nat'd network behind a > nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. > > > However, I can't get any of my phones to Transfer or Blind Transfer.. > > I se

Re: [asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone

On Thu, Jan 7, 2010 at 11:15 AM, William Stillwell (Lists) < william.stillwell-li...@ablebody.net> wrote: > I have several sip stations that on a that are on a nat'd network behind a > nice friend firewall.. no audio path issues, 2 way audio works, > etc,etc,etc. > > > However, I can't get any of

[asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is

Re: [asterisk-users] Please remove me from the mailing list.

Go to this address for information on how to remove yourself:- http://lists.digium.com/mailman/listinfo/asterisk-users -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Dean Sent: 07 January 2010 15:50 To: A

Re: [asterisk-users] Please remove me from the mailing list.

read your posting and it will tell you haw to remove yourself. On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean wrote: > Can I be taken off the mailing list please. > > Thanks. > rick > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.c

[asterisk-users] Please remove me from the mailing list.

Can I be taken off the mailing list please. Thanks. rick ___ -- 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-

Re: [asterisk-users] How to dial a number make two phone Ring at the same time?

On Thu, Jan 7, 2010 at 3:07 PM, Zhang Shukun wrote: > Thank you! > but how can i determine whether ring at the same time or > > alternative ring? > > BTW, the uri > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con It got mistyped or cut, it's http://www.voip-info

Re: [asterisk-users] Zaptel compilation problems: Data Mode!!

On Thu, Jan 07, 2010 at 07:27:00AM -0600, mos...@infolog.mr wrote: > > Nobody can help me on this?? > > -- > > Hi all, > I want to compile zaptel in data mode but i got this errors: > > /usr/src/zaptel-1.4.12.1/kernel/zapte

Re: [asterisk-users] How to dial a number make two phone Ring at the same time?

Thank you! but how can i determine whether ring at the same time or alternative ring? BTW, the uri http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con can't open. Could you paste it again? 2010/1/7 Randy R : > On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun wrote: >>

[asterisk-users] Dialing OutBound SIP trunk using Dial() command

Hello users, i am working on directly calling the numbers from the sip provider of my choice from asterisk using Dial command as follows. extensions.conf [dial-out] exten => _XX,1,NoOp(Dialing out) exten => _XX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

On 01/07/2010 07:21 PM, Olivier wrote: > PS: If you compile Asterisk from source after installing spandsp, > SendFAX and ReceiveFAX would automatically be included. > I opened another thread about that but I doubt that both SendFAX and > ReceiveFAX behave exactly the same (no side effect), no mat

Re: [asterisk-users] How to dial a number make two phone Ring at the same time?

On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun wrote: > hi, > > i want to dial a number to let two phone ring at the same time or > alternative ring, > > how should i configure in asterisk? or how to right the Dialplan code? exten => 12345,1,Dial(${PHONE1}&${PHONE2}) each phone variable is defined

[asterisk-users] How to dial a number make two phone Ring at the same time?

hi, i want to dial a number to let two phone ring at the same time or alternative ring, how should i configure in asterisk? or how to right the Dialplan code? Thanks very much! -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] Zaptel compilation problems: Data Mode!!

Nobody can help me on this?? -- Hi all, I want to compile zaptel in data mode but i got this errors: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ: /usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618:

Re: [asterisk-users] Explain what asterisk.conf's "internal timing" option is

7 jan 2010 kl. 12.00 skrev Olivier: > Hello, > > I've read in Mantis that asterisk.conf's "internal timing" option could > positively impact Asterisk behaviour during faxing > (http://issues.asterisk.org/view.php?id=16374). > Before using it, I would be very pleased to read a line or two about

Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

7 jan 2010 kl. 10.21 skrev Aggio Alberto: > Hi, > I have occasionally experienced the same problem too, and I suspect it was > caused by some spikes in network traffic (e.g. for an intensive file > transfer) that delayed too much SIP OPTION response, so that Asterisk marked > these devices as

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

PS: If you compile Asterisk from source after installing spandsp, SendFAX and ReceiveFAX would automatically be included. I opened another thread about that but I doubt that both SendFAX and ReceiveFAX behave exactly the same (no side effect), no matter the installed spandsp version. I would be ve

Re: [asterisk-users] Really Silly Question From Total Newbie

Hello Tiago, I think that this is the route I will be trying to go as its a proof of concept sort of project. After that - we'll see. Thank you! On Thu, Jan 7, 2010 at 4:43 AM, Tiago Geada wrote: > Hello there! > > If your box has a live Internet connection, then all you need is a sip > provide

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

2010/1/7 Johann Steinwendtner > Olivier schrieb: > > > > > > 2010/1/7 David Backeberg > > > > > > On Wed, Jan 6, 2010 at 6:23 PM, Olivier > > wrote: > > > The second time I'm dialing an internal extension attached to the > >

[asterisk-users] HDLC Receiver overrun

Can anyone shed any light on this error? Will Jan 7 11:03:34 asterisk pppd[9168]: Plugin zaptel.so loaded. Jan 7 11:03:34 asterisk pppd[9168]: Zaptel Plugin Initialized Jan 7 11:03:34 asterisk pppd[9168]: Using zaptel device 'stdin' Jan 7 11:03:34 asterisk pppd[9168]: pppd 2.4.4 started by ro

[asterisk-users] queue and linear strategy

Hello, I've upgraded asterisk to 1.6.0.20 version and found , if I want change queue strategy to linear, I must restart Asterisk: [Jan 7 08:16:10] WARNING[9578]: app_queue.c:1304 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. [Jan 7 08:16:10] WARN

[asterisk-users] Explain what asterisk.conf's "internal timing" option is

Hello, I've read in Mantis that asterisk.conf's "internal timing" option could positively impact Asterisk behaviour during faxing ( http://issues.asterisk.org/view.php?id=16374). Before using it, I would be very pleased to read a line or two about its use. I've read http://www.russellbryant.net/bl

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

Olivier schrieb: > > > 2010/1/7 David Backeberg > > > On Wed, Jan 6, 2010 at 6:23 PM, Olivier > wrote: > > The second time I'm dialing an internal extension attached to the > same > > ReceiveFAX application : >

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

2010/1/7 David Backeberg > On Wed, Jan 6, 2010 at 6:23 PM, Olivier wrote: > > The second time I'm dialing an internal extension attached to the same > > ReceiveFAX application : > > > > 2. sendfax/hylafax/iaxmodem > asterisk > spandsp > > > > In the 2nd case, I've got 3 craches out of

Re: [asterisk-users] Really Silly Question From Total Newbie

Hello there! If your box has a live Internet connection, then all you need is a sip provider. Back to when I lived in the UK, there was this "voipuser.org" which gave me a fixed british number for free, and some outbound call minutes too. I'm sure that if you search around for SIP Providers, you

Re: [asterisk-users] No reply to SIP OPTIONS - sip peers becoming randomly UNREACHABLE

Hi, I have occasionally experienced the same problem too, and I suspect it was caused by some spikes in network traffic (e.g. for an intensive file transfer) that delayed too much SIP OPTION response, so that Asterisk marked these devices as UNREACHABLE; I was able to use the devices too: in fac

Re: [asterisk-users] caller getting cut off intermittently

We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf Thanks for any help John 2010/1/4 John Taylor : > I have recently moved our asterisk server from our LAN to a Debian > Lenny ser

Re: [asterisk-users] DTMF detection on dahdi with b4xxp (again, some more details)

Hi, Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune: > Hi, > > I tried again getting DTMF detection on my ISDN devices with dahdi going > again. I used the channel debug to see whether asterisk sees the frames > and detects them as DTMF. > > Interestingly here's what works: >

Re: [asterisk-users] error compile dahdi with latest kernels.

On Thu, Jan 07, 2010 at 04:19:21PM +0800, james.zhu wrote: > hello, all of users: > there are header files missed when you compile dahdi with kernel-2.6.29 or > 2.6.33. i believe > that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, > opvxa1200.c... > the errors look like these: >

Re: [asterisk-users] How to see STDERR message?

Hello, STDERR goes to the original Asterisk process only, not any "asterisk -r" connections that you may use. If you launch Asterisk in a "screen" like we do, then you can see it and log it in context with when the output is happening. We find it very useful to do it this way. MATT--- On 1/7/10

[asterisk-users] error compile dahdi with latest kernels.

hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: from /usr/src/dahdi-lin

Re: [asterisk-users] How to see STDERR message?

Thank you for you reply? is that mean STDERR couldn't show under Asterisk CLI mode? it's only saved to some file? 2010/1/7 Steve Edwards : > On Thu, 7 Jan 2010, Zhang Shukun wrote: > >> i use agi to send message back to Asterisk by STDERR, but why i could't >> see the message in asterisk CLI? >