Re: [asterisk-users] Digium B410P or Sangoma A502D?
2008/4/24 Patrick <[EMAIL PROTECTED]>: > Hi, > > I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the > specs of various cards I favor the Digium B410P and Sangoma A502D > because of hardware echo cancellation. Does anyone have any experience > with either card, good or bad? Which one would you choose and why? Maybe you should also care about PCI or PCI-E interface. > > > Thanks for your insight. > > Regards, > Patrick > > > ___ > -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
Hello ppl, One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback of gsm files. So scouring the internet gave me the solution of installing ztdummy and loading it as a module. Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill no effect. Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some time now), you don't really need to set anything over there for any zaptel specific compilation? And yes, all the files are gsm files and the codec used for the calls is ulaw. I even tried converting those gsm files to wav using sox and then playing them, but the behaviour is the same. Any ideas anyone.. something I am missing ?? TiA, - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Digium B410P or Sangoma A502D?
2008/4/24 Andres <[EMAIL PROTECTED]>: > You can chose 2/4/6 ports to buy and if you need more > just add remoras up to 24 ports. Is this still usable within 1U server, when you cannot "stack" PCI cards like this xxx xxx but you must align them like this xxx xxx > > > Andres. > > > ___ -- 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] Digium B410P or Sangoma A502D?
2008/4/25 Matt Watson <[EMAIL PROTECTED]>: > I haven;t used any BRI cards but... call me crazy but wouldn;t they still > be using Zaptel (even your sangoma... the script might just be configuring > it for you)... > > and btw, software echo cancel happens in the zaptel kernel driver... I think (but I'm not certain) that it's correct : Digium's B410P are used through chan_misdn. (Please, do not hesitate to correct this) it has nothing to do with the hardware (hence why its a software echo > cancel) > > You also would of had the option of buying HPEC licenses for software echo > cancel from digium for a rather cheap price. This also doesn't apply to chan_misdn hardware ... > > > -- > Matt > > From: [EMAIL PROTECTED] [ > [EMAIL PROTECTED] On Behalf Of Andres [ > [EMAIL PROTECTED] > Sent: Thursday, April 24, 2008 5:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D? > > We have tested both and they work fine. The Sangoma is much easier to > install as it does not depend on any other driver, you just run > 'setup-sangoma' and follow the instructions. You don't have to fiddle > with the linux kernel or zaptel or chan_misdn. It just works. Plus > its more modular. You can chose 2/4/6 ports to buy and if you need more > just add remoras up to 24 ports. The Digium card is fixed to 4 ports, > period. > > Having said that, make sure you stick with the version that has hardware > echo cancel and not even try the other one. We made the mistake of > buying the first time without echo cancel expecting to test the > 'software echo cancel'. But there is no such thing as 'software echo > cancel' on this card. I do not even understand why Sangoma would make a > version without the hardware echo cancel. You get some degree of echo > on practically every call. > > Andres. > > > > Patrick wrote: > > >Hi, > > > >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the > >specs of various cards I favor the Digium B410P and Sangoma A502D > >because of hardware echo cancellation. Does anyone have any experience > >with either card, good or bad? Which one would you choose and why? > > > >Thanks for your insight. > > > >Regards, > >Patrick > > > > > >___ > >-- 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] followme scenarios
Hi All,
I'm tryng to test different scenarios for followme for different users:
[localext]
exten => 101,1,Set(FM = "ALWAYS");
exten => 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101);
exten => 101,n,Hangup
exten => 102,1,Set(FM = "NEVER");
exten => 102,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-102|fm-102);
exten => 102,n,Hangup
exten => 103,1,Set(FM = "WHENBUSY");
exten => 103,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten => 103,n,Hangup
exten => 104,1,Set(FM = "WHENUNAVAILABLE");
exten => 104,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten => 104,n,Hangup
exten => 105,1,Set(FM = "CUSTOM");
exten => 105,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-103|fm-103);
exten => 105,n,Hangup
[macro-dial-ext]
exten => s,1,SetMusicOnHold(${ARG3})
exten => s,n,Dial(${ARG1},5,M(setmusiconhold,${ARG3}))
exten => s,n,GotoIf(FM = "NEVER"|?vm)
exten => s,n,GotoIf(FM = "CUSTOM"|?s-CUSTOM,1)
exten => s,n,GotoIf(FM = "WHENUNAVAILABLE"|?s-CHANUNAVAIL)
exten => s,n,GotoIf(FM = "WHENBUSY"|?s-BUSY)
exten => s-CHANUNAVAIL,1,Followme(${ARG4})
exten => s-BUSY,1,Followme(${ARG4})
exten => s-CUSTOM,1,GotoIftime(17:00-19:00|*|*|*?c-CUSTOM,n)
exten => s-CUSTOM,n,Followme(${ARG4})
exten => s,n,Followme(${ARG4})
exten => s,n(vm),Voicemail([EMAIL PROTECTED]|u)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup
but it just keeps on going to this line
exten => s,n,GotoIf(FM = "NEVER"|?vm)
ami using GotoIf correctly? or am i referring to the FM variable properly? and
is there easier way of doing this? TIA
regards
Ron
-
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Re: [asterisk-users] Drag and Drop transfer application
> > how stable is that? > > The version I used is probably a couple of versions old now, and it > was pretty reliable then. I imagine it would has probably at least > stayed as stabled if not improved a bit. Mmmm. Me talk well english! At the risk of being redundant and wasting list resources, let me try that again... The version I used is probably a couple of versions old now, and it was pretty reliable then. I imagine it has at least stayed as stable, and has probably improved a bit. > > I'm playing with it but so far drag and dropping phone icon to another > phone > > disconnectes the call. > > The setup is not necessarily easy. I spent quite some time with it > before I had everything working the way I wanted. I don't know if > this is still the case, but the developer, Nicolas, is very helpful > and maintains his own mailing list for support issues. And... The developer, Nicolas, is very helpful. I don't know if this is still the case, but he used to maintain his own mailing list for support issues. My lesson: I can't type and take care of a baby at the same time. Oy. - Noah > > > > > > > > > > > > On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins <[EMAIL PROTECTED]> wrote: > > > > > > > > > > > > Al lists wrote: > > > > Hi list, > > > > Any good drag and drop transfer call application for windows based > > > > systems you can advise ? > > > > Something like HUD perhaps? > > > > > > > > > > > > > > Yes. > > > > > > Maestro Control Panel (I authored this one) > > > http://www.datatrakpos.com/pos/datatalk/maestro.aspx. > > > > > > There is also the nice flash based Flash Operator Panel > > > http://www.datatrakpos.com/pos/datatalk/maestro.aspx > > > > > > There a couple of other ones out there too that I thought were nice, but > > can't > > > remember the names. You should be able to find them by gooling for > > "Asterisk > > > Control Panel" or such query. > > > > > > -- > > > > > > Warm Regards, > > > > > > Lee > > > > > > "When my company started out, we were really, really, really, really > > small. > > > Now...we're just really small." > > > > > > > > > > > > > > > ___ > > > -- 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] Drag and Drop transfer application
> any of you guys have used FOP for drag and drop transfer on 30 40 phones > environment? At one point, I used it for about 35 phones (25 users). I had to really do some adjusting to the size of the buttons, but it worked well. I thought it was very useful, as it showed MWI status, and was great for dragging and dropping people into meetme conferences. In the end, though, the receptionist found it easier to just do transfers with the phone buttons, so we abandoned it. YMMV. > how stable is that? The version I used is probably a couple of versions old now, and it was pretty reliable then. I imagine it would has probably at least stayed as stabled if not improved a bit. > I'm playing with it but so far drag and dropping phone icon to another phone > disconnectes the call. The setup is not necessarily easy. I spent quite some time with it before I had everything working the way I wanted. I don't know if this is still the case, but the developer, Nicolas, is very helpful and maintains his own mailing list for support issues. - Noah > > > > > > On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins <[EMAIL PROTECTED]> wrote: > > > > > > > > Al lists wrote: > > > Hi list, > > > Any good drag and drop transfer call application for windows based > > > systems you can advise ? > > > Something like HUD perhaps? > > > > > > > > > > Yes. > > > > Maestro Control Panel (I authored this one) > > http://www.datatrakpos.com/pos/datatalk/maestro.aspx. > > > > There is also the nice flash based Flash Operator Panel > > http://www.datatrakpos.com/pos/datatalk/maestro.aspx > > > > There a couple of other ones out there too that I thought were nice, but > can't > > remember the names. You should be able to find them by gooling for > "Asterisk > > Control Panel" or such query. > > > > -- > > > > Warm Regards, > > > > Lee > > > > "When my company started out, we were really, really, really, really > small. > > Now...we're just really small." > > > > > > > > > > ___ > > -- 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] No DTMF on Sip Connection between two asterisk boxes?
Hi Olle - > Actually, there's a large difference between an IAX2 trunk and an IAX2 > connection. > > The IAX2 trunk multiplexes multiple media streams in one UDP packet, > therefore you can call it trunking. In order for this to work, you > need to enable a zaptel timer source in your system. > > As Eric say, there's no trunking support similar to IAX2 trunks in the > SIP channel driver. > > Semantics, but important in this case. :-) Well, I stand corrected, and straight from the SIP-Lord's* fingers. I have adjusted the subject of this thread accordingly. I guess I was thinking of word "trunk" colloquially, as in a something that connects calls from multiple devices to another location. Anyhoo, I'll go ahead and ask Digium support, but if anyone here has any insight, please let me know. Since I changed the thread subject, I'll repost the original question: For the first time, I'm setting up SIP connections between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. I've tried using inband, rfc2833 and auto, and none of them work. Maybe I'm missing something obvious? Here's my config: Asterisk1 (1.2.18): sip.conf [129trunk551] type=friend secret= username=129trunk551 host=xxx.xxx.xxx.xxx context=phones dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Asterisk2 (ABE revC): sip.conf [129trunk551] type=friend secret=*** username=129trunk551 host=yyy.yyy.yyy.yyy context=default dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Thanks! Noah * In the asterisk universe, SIP-Lords are the good guys ;-) ___ -- 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] No DTMF on Sip Trunk?
No, it is not the same thing. An IAX2 Trunk is a version of IAX2 that puts audio from multiple calls between the same two servers into a single UDP packet. Fewer packets need to be sent so you use the bandwidth much more efficiency because you don't have the packet header overhead. SIP does nothing similar. Noah Miller wrote: >> For ABE support you really should contact Digium. BTW, there is no such >> thing as a "sip trunk". It's a sip peer or connection or account. > > Semantics. IAX connections between two asterisk boxes are > often called IAX trunks. This is the same thing in SIP > flavor. > > Also, no offense against Digium support, but the list actually > responds more quickly at this point. I think the Digium support staff > are in a situation of high demand and short staffing. ___ -- 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] Drag and Drop transfer application
any of you guys have used FOP for drag and drop transfer on 30 40 phones environment? how stable is that? I'm playing with it but so far drag and dropping phone icon to another phone disconnectes the call. On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins <[EMAIL PROTECTED]> wrote: > Al lists wrote: > > Hi list, > > Any good drag and drop transfer call application for windows based > > systems you can advise ? > > Something like HUD perhaps? > > > > > > Yes. > > Maestro Control Panel (I authored this one) > http://www.datatrakpos.com/pos/datatalk/maestro.aspx. > > There is also the nice flash based Flash Operator Panel > http://www.datatrakpos.com/pos/datatalk/maestro.aspx > > There a couple of other ones out there too that I thought were nice, but > can't > remember the names. You should be able to find them by gooling for > "Asterisk > Control Panel" or such query. > > -- > > Warm Regards, > > Lee > > "When my company started out, we were really, really, really, really small. > Now...we're just really small." > > ___ > -- 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] Digium B410P or Sangoma A502D?
The Sangoma kernel drivers are different than Zaptel, while running the install script you are asked if you would like to generate the Zaptel configs but it is not required, you must also run wancfg to configure the cards beyond the Zaptel configs. The Sangoma drivers kind of run on top of the Zaptel. It seems that the newest wanpipe drivers and Zaptel 1.4 work without the D chan patch which is very nice IMO, I hate patches. I have run the BRIStuff install and it has tons of patches! Kind of scary but it works for it's purpose. I have only done BRI once but there was absolutely no echo by simply setting echocancel=yes, echocancelwhenbridged=no. I hear "might as well get the hardware EC board" quite a bit, but on all the many dozens of PRIs I have installed, software EC has been adequate (if needed at all). It would have meant quite a bit of wasted money that was better spent on a nice 48 port gigabit switch. I have tested both ways (hardware vs. software), no difference really (Sangoma). Sangoma actually sent me one of each before purchasing seven quad cards to test if hardware EC was going to be required for one deployment. I returned the hardware EC card and ordered seven quad PRI cards. Maybe I am just lucky or have not had enough exposure to BRI but ISDN is ISDN, right (it really is a question, I don't know)? Now on analog, that is a horse of a different color, also the phone on either side, but especially your side can be the culprit (older Grandstream for one) Polycom seems to eliminate much of this. Thanks, Steve Totaro On Thu, Apr 24, 2008 at 7:50 PM, Matt Watson <[EMAIL PROTECTED]> wrote: > I haven;t used any BRI cards but... call me crazy but wouldn;t they still be > using Zaptel (even your sangoma... the script might just be configuring it > for you)... > > and btw, software echo cancel happens in the zaptel kernel driver... it has > nothing to do with the hardware (hence why its a software echo cancel) > > You also would of had the option of buying HPEC licenses for software echo > cancel from digium for a rather cheap price. > > -- > Matt > > From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL > PROTECTED] > Sent: Thursday, April 24, 2008 5:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D? > > > > We have tested both and they work fine. The Sangoma is much easier to > install as it does not depend on any other driver, you just run > 'setup-sangoma' and follow the instructions. You don't have to fiddle > with the linux kernel or zaptel or chan_misdn. It just works. Plus > its more modular. You can chose 2/4/6 ports to buy and if you need more > just add remoras up to 24 ports. The Digium card is fixed to 4 ports, > period. > > Having said that, make sure you stick with the version that has hardware > echo cancel and not even try the other one. We made the mistake of > buying the first time without echo cancel expecting to test the > 'software echo cancel'. But there is no such thing as 'software echo > cancel' on this card. I do not even understand why Sangoma would make a > version without the hardware echo cancel. You get some degree of echo > on practically every call. > > Andres. > > > > Patrick wrote: > > >Hi, > > > >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the > >specs of various cards I favor the Digium B410P and Sangoma A502D > >because of hardware echo cancellation. Does anyone have any experience > >with either card, good or bad? Which one would you choose and why? > > > >Thanks for your insight. > > > >Regards, > >Patrick > > > > > >___ > >-- 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
Re: [asterisk-users] Digium B410P or Sangoma A502D?
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be using Zaptel (even your sangoma... the script might just be configuring it for you)... and btw, software echo cancel happens in the zaptel kernel driver... it has nothing to do with the hardware (hence why its a software echo cancel) You also would of had the option of buying HPEC licenses for software echo cancel from digium for a rather cheap price. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL PROTECTED] Sent: Thursday, April 24, 2008 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D? We have tested both and they work fine. The Sangoma is much easier to install as it does not depend on any other driver, you just run 'setup-sangoma' and follow the instructions. You don't have to fiddle with the linux kernel or zaptel or chan_misdn. It just works. Plus its more modular. You can chose 2/4/6 ports to buy and if you need more just add remoras up to 24 ports. The Digium card is fixed to 4 ports, period. Having said that, make sure you stick with the version that has hardware echo cancel and not even try the other one. We made the mistake of buying the first time without echo cancel expecting to test the 'software echo cancel'. But there is no such thing as 'software echo cancel' on this card. I do not even understand why Sangoma would make a version without the hardware echo cancel. You get some degree of echo on practically every call. Andres. Patrick wrote: >Hi, > >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the >specs of various cards I favor the Digium B410P and Sangoma A502D >because of hardware echo cancellation. Does anyone have any experience >with either card, good or bad? Which one would you choose and why? > >Thanks for your insight. > >Regards, >Patrick > > >___ >-- 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] X101P [Re: buying cards from pakistan]
Tzafrir, I'm sorry: I sent my previous message at your private address, it was a mistake :-(sorry :-) >Tzafrir Cohen > icq#16849755 jabber:[EMAIL PROTECTED] > +972-50-7952406 mailto:[EMAIL PROTECTED] > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir best regards, giuliano curti ___ -- 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] No DTMF on Sip Trunk?
> Actually, Digium Support has been quite responsive in recent weeks, as > noted on this list 2 weeks ago: > > http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html > > We strive to be as responsive as we can, and have had some success on > this front recently. Please give us a chance! Thanks Kenny! I don't mean to disparage you folks. You've always been extremely knowledgeable and courteous. Glad to see you get some praise. I just had a simple little question, and I thought I'd ask on the list to see if anyone else had seen this before. - Noah ___ -- 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] No DTMF on Sip Trunk?
> Forwarded Message > From: Noah Miller <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] No DTMF on Sip Trunk? > Date: Thu, 24 Apr 2008 17:01:18 -0400 > > >> For ABE support you really should contact Digium. BTW, there is no such >> thing as a "sip trunk". It's a sip peer or connection or account. >> > > Semantics. IAX connections between two asterisk boxes are > often called IAX trunks. This is the same thing in SIP > flavor. > > Also, no offense against Digium support, but the list actually > responds more quickly at this point. I think the Digium support staff > are in a situation of high demand and short staffing. > > > - Noah > Actually, Digium Support has been quite responsive in recent weeks, as noted on this list 2 weeks ago: http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html We strive to be as responsive as we can, and have had some success on this front recently. Please give us a chance! Noah, if you have a specific support experience where we weren't as responsive as we could have been, please contact me off-list to discuss. I want to hear about it! ~Kenny Shumard Digium Technical Support Manager ___ -- 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] Disable transfer on all calls
Most times it's easier to find something in google, than in your own
computer :)
2008/4/25, Eric Wieling <[EMAIL PROTECTED]>:
>
> In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
> in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
>
> The "doc" directory is the only official source of documentation for
> Asterisk that I am aware of. Read it.
>
>
> [EMAIL PROTECTED] wrote:
> > Dinesh Nair пишет:
> >> On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
> >>
> >>> The best option is to put a SIP Proxy in front of your Asterisk sever
> >>> and block REFER requests.
> >>>
> >> or just comment out the block in chan_sip.c which handles the refers.
> >>
> >>
> >
> > Thanks to your answers, but i found more beautiful way to do this -
> > there is some system variable __TRANSFER_CONTEXT, which defines context
> > to handle the transfered number, so you can create a new context and
> > there you can do anything with transfered call - i just hang it up.
> >
> > It's really strange that this is in fact undocumented function - you can
> > find it only in comments on wiki at voip-info.org. Man there said that
> > he found this variable while hacking source code of asterisk:
> >
> > $ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/
> > /usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =
> > pbx_builtin_getvar_helper(sip_pvt->owner, "TRANSFER_CONTEXT");
> > /usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}
> > Context for transferred calls
> > /usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did
> > not use the TRANSFER_CONTEXT
> > /usr/src/asterisk-1.2.15/res/res_features.c: if
> > (!(transferer_real_context = pbx_builtin_getvar_helper(transferee,
> > "TRANSFER_CONTEXT")) &&
> > /usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =
> > pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT"))) {
> > /usr/src/asterisk-1.2.15/res/res_features.c: if
> > (!(transferer_real_context=pbx_builtin_getvar_helper(transferee,
> > "TRANSFER_CONTEXT")) &&
> > /usr/src/asterisk-1.2.15/res/res_features.c:
> > !(transferer_real_context=pbx_builtin_getvar_helper(transferer,
> > "TRANSFER_CONTEXT"))) {
> >
> >
>
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>
>
> --
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> T-1, PRI, Frame Relay, Linux, and network design. Based near
> Birmingham, AL. Now accepting clients worldwide.
>
>
>
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Re: [asterisk-users] No DTMF on Sip Trunk?
24 apr 2008 kl. 23.01 skrev Noah Miller: >> For ABE support you really should contact Digium. BTW, there is no >> such >> thing as a "sip trunk". It's a sip peer or connection or account. > > Semantics. IAX connections between two asterisk boxes are > often called IAX trunks. This is the same thing in SIP > flavor. > > Also, no offense against Digium support, but the list actually > responds more quickly at this point. I think the Digium support staff > are in a situation of high demand and short staffing. > >>> Actually, there's a large difference between an IAX2 trunk and an IAX2 connection. The IAX2 trunk multiplexes multiple media streams in one UDP packet, therefore you can call it trunking. In order for this to work, you need to enable a zaptel timer source in your system. As Eric say, there's no trunking support similar to IAX2 trunks in the SIP channel driver. Semantics, but important in this case. :-) /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * The Asterisk SIP Masterclass in Barcelona, May 5-9 - REGISTER now! ___ -- 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] Digium B410P or Sangoma A502D?
We have tested both and they work fine. The Sangoma is much easier to install as it does not depend on any other driver, you just run 'setup-sangoma' and follow the instructions. You don't have to fiddle with the linux kernel or zaptel or chan_misdn. It just works. Plus its more modular. You can chose 2/4/6 ports to buy and if you need more just add remoras up to 24 ports. The Digium card is fixed to 4 ports, period. Having said that, make sure you stick with the version that has hardware echo cancel and not even try the other one. We made the mistake of buying the first time without echo cancel expecting to test the 'software echo cancel'. But there is no such thing as 'software echo cancel' on this card. I do not even understand why Sangoma would make a version without the hardware echo cancel. You get some degree of echo on practically every call. Andres. Patrick wrote: >Hi, > >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the >specs of various cards I favor the Digium B410P and Sangoma A502D >because of hardware echo cancellation. Does anyone have any experience >with either card, good or bad? Which one would you choose and why? > >Thanks for your insight. > >Regards, >Patrick > > >___ >-- 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] No DTMF on Sip Trunk?
> For ABE support you really should contact Digium. BTW, there is no such > thing as a "sip trunk". It's a sip peer or connection or account. Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor. Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. - Noah > > > > Noah Miller wrote: > > Hi Jared - > > > >> > For the first time, I'm setting up SIP trunking between two asterisk > >> > boxes. The calls themselves work fine, but I'm not able to get DTMF > >> > working. > >> > >> If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it > >> appears that you are), you'll need to set "rfc2833compensate=yes" in the > >> peer or friend section of sip.conf on the Asterisk 1.4 box. > > > > Unfortunately, this didn't work. Maybe rfc2833compensate isn't > > available in ABE? > > > > I think this may require inband signalling anyway, as we'll require > > non-sip (zap) devices to be able to use these sip trunks and enter > > DTMF. > > > > Any other ideas? > > > > Thanks! > > Noah > > > > > ___ > > -- 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 > > > > > > -- > Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, > T-1, PRI, Frame Relay, Linux, and network design. Based near > Birmingham, AL. Now accepting clients worldwide. > > > > ___ > -- 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] Full queue issues
Hello everyone.
I got a little problem in here: I want to set up a queue so that if anything of
these happens:
a) No agents logged in
b) All agents busy
Then the user gets diverted somewhere. I used this (for testing purposes only,
of course):
exten => 7080,1,Answer()
exten => 7080,n,Queue(teste)
exten => 7080,n,Goto(${QUEUESTATUS})
exten => 7080,n(ERROR),NoOp(${QUEUESTATUS})
exten => 7080,n,Hangup()
exten => 7080,n(LEAVEEMPTY),Goto(ERROR)
exten => 7080,n(TIMEOUT),Goto(ERROR)
exten => 7080,n(JOINUNAVAIL),Goto(ERROR)
exten => 7080,n(LEAVEUNAVAIL),Goto(ERROR)
exten => 7080,n(JOINEMPTY),Goto(ERROR)
exten => 7080,n(TIMEOUT),Goto(ERROR)
exten => *210,1,AddQueueMember(teste,SIP/${CALLERID(num)})
exten => *210,n,UserEvent(RefreshQueue)
exten => *210,n,Playback(agent-loginok)
exten => *220,1,RemoveQueueMember(teste,SIP/${CALLERID(num)})
exten => *220,n,UserEvent(RefreshQueue)
exten => *220,n,Playback(agent-loggedoff)
In queues.conf:
[teste]
strategy=roundrobin
music=default
timeout=10
retry=0
maxlen=1
ringinuse=no
leavewhenempty=strict
joinempty=strict
Then I have those scenarios:
a) There is no agents logged in, a call tries to enter the queue, the
${QUEUESTATUS} variable is set to LEAVEEMPTY and the call is disconnected.
Everything fine in here.
b) There is only one agent logged in, he's in a call (InUse), the call enters
the queue and stays there. I would like the call NOT to enter the queue and the
${QUEUESTATUS} variable to be set to something different.
Am I missing something or it's just not possible? I'm using SIP phones for the
agents and Asterisk 1.4.15.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
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[asterisk-users] ATA FXO / FXS - can forward to sip ?
Hi All, Quick question. We have a customer with a T1 located in their data center, and then one TDM card for local calls at their remote offices. We would like to remove the local PBX and TDM card and have them register directly to the main server. For the remote office, that still uses one local telephone number over analogue, we were thinking of getting an ATA device with two FXS and one FXO. The FXO would connect directly to Bell, and the FXS would go to an internal fax machine (outgoing only), and one internal analogue phone. Now, our question is. Since the IVR resides on the server in the datacenter, does anyone know of any ATA devices that will let us forward all calls, over sip (or iax) to the pbx to hit the IVR? We basically only need the local office number for emergencies, and when callers hit it, they should usually get the IVR, unless power is out, in which case the regular analogue phone would work. Anyone have any ideas? Thanks, Matt ___ -- 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] Quality problems with ISDN PRI
On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote: > There are much better solutions than doing a RAM drive. While it may > be stable (not in my experience, I advise using different servers for > different tasks (with redundancy obviously). A phone switch should be > just that, a recording server should also be just that (in demanding > environments). That would be fine, if Asterisk was capable of buffering recording writes, but I'm told it's not; the I/O involved in getting that recording data off the box in real time is probably worse than that of putting it onto disk -- disks are usually higher bandwidth channels than network adapters. For permanent storage, certainly, the recordings should be moved to another box, and that's how we do it here. Cheers, -- jr '44 byte chunks. Is someone an ATM fan?' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Disable transfer on all calls
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
The "doc" directory is the only official source of documentation for
Asterisk that I am aware of. Read it.
[EMAIL PROTECTED] wrote:
> Dinesh Nair пишет:
>> On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
>>
>>> The best option is to put a SIP Proxy in front of your Asterisk sever
>>> and block REFER requests.
>>>
>> or just comment out the block in chan_sip.c which handles the refers.
>>
>>
>
> Thanks to your answers, but i found more beautiful way to do this -
> there is some system variable __TRANSFER_CONTEXT, which defines context
> to handle the transfered number, so you can create a new context and
> there you can do anything with transfered call - i just hang it up.
>
> It's really strange that this is in fact undocumented function - you can
> find it only in comments on wiki at voip-info.org. Man there said that
> he found this variable while hacking source code of asterisk:
>
> $ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/
> /usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =
> pbx_builtin_getvar_helper(sip_pvt->owner, "TRANSFER_CONTEXT");
> /usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}
> Context for transferred calls
> /usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did
> not use the TRANSFER_CONTEXT
> /usr/src/asterisk-1.2.15/res/res_features.c: if
> (!(transferer_real_context = pbx_builtin_getvar_helper(transferee,
> "TRANSFER_CONTEXT")) &&
> /usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =
> pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT"))) {
> /usr/src/asterisk-1.2.15/res/res_features.c: if
> (!(transferer_real_context=pbx_builtin_getvar_helper(transferee,
> "TRANSFER_CONTEXT")) &&
> /usr/src/asterisk-1.2.15/res/res_features.c:
> !(transferer_real_context=pbx_builtin_getvar_helper(transferer,
> "TRANSFER_CONTEXT"))) {
>
>
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T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
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Re: [asterisk-users] No DTMF on Sip Trunk?
For ABE support you really should contact Digium. BTW, there is no such thing as a "sip trunk". It's a sip peer or connection or account. Noah Miller wrote: > Hi Jared - > >> > For the first time, I'm setting up SIP trunking between two asterisk >> > boxes. The calls themselves work fine, but I'm not able to get DTMF >> > working. >> >> If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it >> appears that you are), you'll need to set "rfc2833compensate=yes" in the >> peer or friend section of sip.conf on the Asterisk 1.4 box. > > Unfortunately, this didn't work. Maybe rfc2833compensate isn't > available in ABE? > > I think this may require inband signalling anyway, as we'll require > non-sip (zap) devices to be able to use these sip trunks and enter > DTMF. > > Any other ideas? > > Thanks! > Noah > > ___ > -- 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 > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] GoToIfTime problem
Lee Jenkins wrote:
> -- Executing GotoIfTime("Zap/3-1",
> "08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1")
>
Too many pipes. Mine is:
GotoIfTime(00:00-07:50|mon-fri|*|*?auto-paging,s,1)
Doug
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Re: [asterisk-users] GoToIfTime problem
Lee Jenkins wrote:
> I'm having a problem at a custom site where GotoIfTime doesn't seem to be
> working for some reason. I had putty running and logging CLI output and
> below
> is the call data:
>
> -- Executing Answer("Zap/3-1", "") in new stack
> -- Executing Ringing("Zap/3-1", "") in new stack
> -- Executing Wait("Zap/3-1", "0") in new stack
> -- Executing SetMusicOnHold("Zap/3-1", "default") in new stack
> -- Executing Goto("Zap/3-1", "check_time|s|1") in new stack
> -- Goto (check_time,s,1)
> -- Executing GotoIf("Zap/3-1", "0?set_no_callerid|s|1") in new stack
> -- Executing NoOp("Zap/3-1", "CallerID: 443866 Cell Phone MD") in new
> stack
> -- Executing GotoIfTime("Zap/3-1",
> "08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1")
> in new stack
> -- Executing Goto("Zap/3-1", "after_hours|s|1") in new stack
> -- Goto (after_hours,s,1)
>
Never mind, the problem turned out to be between the back of the chair and the
keyboard.
Sorry for the false alarm.
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Lee
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Re: [asterisk-users] GoToIfTime problem
On 15:43, Thu 24 Apr 08, Lee Jenkins wrote:
>
> I'm having a problem at a custom site where GotoIfTime doesn't seem to be
> working for some reason. I had putty running and logging CLI output and
> below
> is the call data:
>
> -- Executing Answer("Zap/3-1", "") in new stack
> -- Executing Ringing("Zap/3-1", "") in new stack
> -- Executing Wait("Zap/3-1", "0") in new stack
> -- Executing SetMusicOnHold("Zap/3-1", "default") in new stack
> -- Executing Goto("Zap/3-1", "check_time|s|1") in new stack
> -- Goto (check_time,s,1)
> -- Executing GotoIf("Zap/3-1", "0?set_no_callerid|s|1") in new stack
> -- Executing NoOp("Zap/3-1", "CallerID: 443866 Cell Phone MD") in new
> stack
> -- Executing GotoIfTime("Zap/3-1",
> "08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1")
> in new stack
> -- Executing Goto("Zap/3-1", "after_hours|s|1") in new stack
> -- Goto (after_hours,s,1)
>
> This call came in at about 3:10 PM EDT today (Thursday). I did a "date"
> command
> at the linux prompt and the date and time of the computer is set correctly.
>
> Now, I have had problem with this particular computer in that the date/time
> gets
> changed somehow, although I'm not sure exactly how. I've changed it back
> several times using the commands (copied from command line history):
>
> # date -s "23 APR 2008 1:42:00"
> # hwclock --utc --systohc
Install ntp so it will sync with the internet all the time.
It's the first package I install on servers, no matter if it's brandnew
or not-so-brandnew hardware.
Time IS important for a lot of applications.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
"Why is it drug addicts and computer aficionados are both called users?"
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Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted
> Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of > the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, > and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4 > than the original Steve driver in 1.2, and with better sound under heavy > loads. > The Asutunicall page can be found here: > http://www.moythreads.com/astunicall/ Hum, wonder who this moy is hey wait, that's me! . Even when is in my plans to keep giving general maintenance to chan_unicall, my long term plan is to leave R2 support into chan_zap, so I would recommend to all users to try chan_zap R2 support, the more users we get the faster the driver will be stable enough to replace chan_unicall, the less headaches you will have (I hope). - Moy or Moisés Silva, same shit :-) -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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] GoToIfTime problem
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and below
is the call data:
-- Executing Answer("Zap/3-1", "") in new stack
-- Executing Ringing("Zap/3-1", "") in new stack
-- Executing Wait("Zap/3-1", "0") in new stack
-- Executing SetMusicOnHold("Zap/3-1", "default") in new stack
-- Executing Goto("Zap/3-1", "check_time|s|1") in new stack
-- Goto (check_time,s,1)
-- Executing GotoIf("Zap/3-1", "0?set_no_callerid|s|1") in new stack
-- Executing NoOp("Zap/3-1", "CallerID: 443866 Cell Phone MD") in new
stack
-- Executing GotoIfTime("Zap/3-1", "08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1")
in new stack
-- Executing Goto("Zap/3-1", "after_hours|s|1") in new stack
-- Goto (after_hours,s,1)
This call came in at about 3:10 PM EDT today (Thursday). I did a "date"
command
at the linux prompt and the date and time of the computer is set correctly.
Now, I have had problem with this particular computer in that the date/time
gets
changed somehow, although I'm not sure exactly how. I've changed it back
several times using the commands (copied from command line history):
# date -s "23 APR 2008 1:42:00"
# hwclock --utc --systohc
I'm still quite the linux noob, so it could be something I'm doing wrong
although it seems doubtful since everything was working until recently.
Maybe the computer's clock battery is screwed?
Thank you,
--
Warm Regards,
Lee
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[asterisk-users] No CallerID Transfer Problem
Came upon a problem today that I thought I'd see if it's by design, if
I'm missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with
others, but for some reason when they would transfer an outside call
anywhere it would pause for a few seconds and hang up the line.
Well, after spending most of the day on it, it turns out it's because
they don't have callerID on the PSTN lines coming in through zaptel. My
first thought was, set "usecallerid=no" and all would be well, but this
didn't do any good. After playing a bit longer I just set the
following:
exten => 900,2,set(CALLERID(num)="606-555-1212")
exten => 900,3,set(CALLERID(name)="Outside Call")
exten =>
900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
Now all works well.
So is there another option somewhere to keep asterisk from killing a
transfer without callerid? This happened on both 1.4.17 & 1.4.18.1.
Thanks,
Ken
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Re: [asterisk-users] help...i cant do more...
On Thu, Apr 24, 2008 at 11:23:21AM -0700, Steve Edwards wrote:
> On Thu, 24 Apr 2008, Bruno Pereira wrote:
>
> > ssh etx9 'sudo /etc/init.d/asterisk start'
> > [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start'
> > start ini
> > Starting asterisk: [ OK ]
> > decrease the verbosity level to zero: OK
> > start fim
> >
> > and just stays there, like waiting for something.
>
> Sudo recently (?) added a new parameter in /etc/sudoers that caused me a
> lot of grief. Comment out "Defaults requiretty" and see if it helps.
>
> Also, your custom /etc/init.d/asterisk script ("start fim?") may not be
> redirecting stdin/stdout/stderr correctly.
Or Asterisk may not be closing file descriptors properly?
I have such an issue with some scripts getting hanged on installation of
the package asterisk on Debian until I restart Asterisk.
--
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+972-50-7952406 mailto:[EMAIL PROTECTED]
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Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4 than the original Steve driver in 1.2, and with better sound under heavy loads. The Asutunicall page can be found here: http://www.moythreads.com/astunicall/ Guillermo > Date: Thu, 24 Apr 2008 12:16:42 -0500 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted > >> Way more handy and will be much more reliable too. Steve Underwood did a >> great job implemeting it, but as far as I know the code isn't actively >> maintained anymore. Of course your implementation of MFC/R2 will take a >> while to become stable, but hey -- it's a start. > > Agreed. > >> Russel pointed some licensing stuff related to the Digivoice drivers. >> Please listen to him on that, I had no idea of that kind of complication. If >> your >> implementation of MFC/R2 can't be integrated in Zaptel, then it's no much >> better than Unicall. > You mean chan_zap/zapata (zaptel is the kernel code). I had already > discussed with Russell the licensing. He just got confused for a > second because he forgot LGPL code requires both license files (GPL > and LGPL). > >> I guess if you look at Digivoice's code to figure out how it works and then >> write your own code, there will be no licensing issues. But that's just a >> guess, >> Russel will need to clarify it. > I did not discussed that with Russell, but I will. In the meantime, > since I am aware of the licensing concerns I have not even looked at > that code :-) > > I would like to test the BR variant the next week, I will contact you > off-list to see if we can meet via IM. > > - Moisés Silva > > -- > "I do not agree with what you have to say, but I'll defend to the > death your right to say it." Voltaire > > ___ > -- 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 _ Send funny voice messages packed with tidbits from MSN. Everyone wants to be ready. http://www.noonewantstolookdumb.com?OCID=T001MSN54N1613A ___ -- 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] help...i cant do more...
On Thu, 24 Apr 2008, Bruno Pereira wrote:
> ssh etx9 'sudo /etc/init.d/asterisk start'
> [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start'
> start ini
> Starting asterisk: [ OK ]
> decrease the verbosity level to zero: OK
> start fim
>
> and just stays there, like waiting for something.
Sudo recently (?) added a new parameter in /etc/sudoers that caused me a
lot of grief. Comment out "Defaults requiretty" and see if it helps.
Also, your custom /etc/init.d/asterisk script ("start fim?") may not be
redirecting stdin/stdout/stderr correctly.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] G723 pass thru
hi, thanks for replying guys, I have a digium transcoder card installed and its running on mixed mode. The softphone I have, is using g723.1 6.3k while the transcoder card is using g723.1 5.3k...so it has different payload size..FYI im using softphone from Adore. The guy from the Adore support told me to use pass-through. cheers, Aby Azid On Thu, Apr 24, 2008 at 10:42 PM, Anthony Francis <[EMAIL PROTECTED]> wrote: > More importantly, for it to "pass-through" you need something that > processes g723 on the other end. If Asterisk is terminating the call by > handing it off to the PSTN or to another phone that does not do g723 > then Asterisk must transcode and that requires the license. > > Eric Wieling wrote: > > allow=g723.1 or allow=g723 (I don't remember which). > > > > aby azid wrote: > > > >> Hi, > >> > >> I have softphone with a g723 codec, my question is how do i set it as > Pass > >> thru in Asterisk? > >> > > > > > > > > -- > Thank you and have any kind of day you want, > > Anthony Francis > > > ___ > -- 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] help...i cant do more...
On Thu, Apr 24, 2008 at 05:01:53PM +0100, Bruno Pereira wrote: > Hi... > Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the > asterisk version is 1.4.18. > If in the machine is all ok, i can stop start the asterisk service no prob, > my problem is when in another server (in my case, debian etch 4) using the > ssh the stop service is ok, but the start service dosend finalise. > Like this: > ssh etx9 'sudo /etc/init.d/asterisk stop' > [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk stop' > Shutting down asterisk: [ OK ] > stop > > ssh etx9 'sudo /etc/init.d/asterisk start' > [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start' > start ini > Starting asterisk: [ OK ] > decrease the verbosity level to zero: OK > start fim > > and just stays there, like waiting for something. > So if someone can help, please do i dont have any more ideias. For starters, trace the init.d script to see where it is hung: sh -x /etc/init.d/asterisk start (As root or from sudo) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] No DTMF on Sip Trunk?
Hi Jared - > > For the first time, I'm setting up SIP trunking between two asterisk > > boxes. The calls themselves work fine, but I'm not able to get DTMF > > working. > > If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it > appears that you are), you'll need to set "rfc2833compensate=yes" in the > peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
> Way more handy and will be much more reliable too. Steve Underwood did a > great job implemeting it, but as far as I know the code isn't actively > maintained anymore. Of course your implementation of MFC/R2 will take a while > to become stable, but hey -- it's a start. Agreed. > Russel pointed some licensing stuff related to the Digivoice drivers. Please > listen to him on that, I had no idea of that kind of complication. If your > implementation of MFC/R2 can't be integrated in Zaptel, then it's no much > better than Unicall. You mean chan_zap/zapata (zaptel is the kernel code). I had already discussed with Russell the licensing. He just got confused for a second because he forgot LGPL code requires both license files (GPL and LGPL). > I guess if you look at Digivoice's code to figure out how it works and then > write your own code, there will be no licensing issues. But that's just a > guess, > Russel will need to clarify it. I did not discussed that with Russell, but I will. In the meantime, since I am aware of the licensing concerns I have not even looked at that code :-) I would like to test the BR variant the next week, I will contact you off-list to see if we can meet via IM. - Moisés Silva -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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] Digium B410P or Sangoma A502D?
I have a box running a TE410P with echo cancelling and it works like a charm. Set up once, forget about it. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Patrick" <[EMAIL PROTECTED]> escreveu: > Hi, > > I need to setup an Asterisk box with 4x ISDN BRI links. Looking at > the > specs of various cards I favor the Digium B410P and Sangoma A502D > because of hardware echo cancellation. Does anyone have any > experience > with either card, good or bad? Which one would you choose and why? > > Thanks for your insight. > > Regards, > Patrick > > > ___ > -- 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] help...i cant do more...
Hi... Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the asterisk version is 1.4.18. If in the machine is all ok, i can stop start the asterisk service no prob, my problem is when in another server (in my case, debian etch 4) using the ssh the stop service is ok, but the start service dosend finalise. Like this: ssh etx9 'sudo /etc/init.d/asterisk stop' [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk stop' Shutting down asterisk: [ OK ] stop ssh etx9 'sudo /etc/init.d/asterisk start' [EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start' start ini Starting asterisk: [ OK ] decrease the verbosity level to zero: OK start fim and just stays there, like waiting for something. So if someone can help, please do i dont have any more ideias. Thanks Bruno Pereira ___ -- 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] Digium B410P or Sangoma A502D?
Hi, I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the specs of various cards I favor the Digium B410P and Sangoma A502D because of hardware echo cancellation. Does anyone have any experience with either card, good or bad? Which one would you choose and why? Thanks for your insight. Regards, Patrick ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
> > Hello Moisés, thanks for your effort on this! I would love to use > Digium cards for MFC/R2 signalling in the future. > Currently you can use Digium cards with Unicall :-) , tho, having > MFC/R2 on chan_zap is more handy. Way more handy and will be much more reliable too. Steve Underwood did a great job implemeting it, but as far as I know the code isn't actively maintained anymore. Of course your implementation of MFC/R2 will take a while to become stable, but hey -- it's a start. > > I added some info you might like in the bugtracker, you might take > a look at it. > I will, thanks! Russel pointed some licensing stuff related to the Digivoice drivers. Please listen to him on that, I had no idea of that kind of complication. If your implementation of MFC/R2 can't be integrated in Zaptel, then it's no much better than Unicall. I guess if you look at Digivoice's code to figure out how it works and then write your own code, there will be no licensing issues. But that's just a guess, Russel will need to clarify it. You can count on me for any testing needed on the BR variant. ___ -- 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] Disable transfer on all calls
Dinesh Nair пишет:
> On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
>
>> The best option is to put a SIP Proxy in front of your Asterisk sever
>> and block REFER requests.
>>
>
> or just comment out the block in chan_sip.c which handles the refers.
>
>
Thanks to your answers, but i found more beautiful way to do this -
there is some system variable __TRANSFER_CONTEXT, which defines context
to handle the transfered number, so you can create a new context and
there you can do anything with transfered call - i just hang it up.
It's really strange that this is in fact undocumented function - you can
find it only in comments on wiki at voip-info.org. Man there said that
he found this variable while hacking source code of asterisk:
$ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/
/usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =
pbx_builtin_getvar_helper(sip_pvt->owner, "TRANSFER_CONTEXT");
/usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}
Context for transferred calls
/usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did
not use the TRANSFER_CONTEXT
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context = pbx_builtin_getvar_helper(transferee,
"TRANSFER_CONTEXT")) &&
/usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =
pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT"))) {
/usr/src/asterisk-1.2.15/res/res_features.c: if
(!(transferer_real_context=pbx_builtin_getvar_helper(transferee,
"TRANSFER_CONTEXT")) &&
/usr/src/asterisk-1.2.15/res/res_features.c:
!(transferer_real_context=pbx_builtin_getvar_helper(transferer,
"TRANSFER_CONTEXT"))) {
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[asterisk-users] IAX issues with 1.4.19.1
I upgraded one of our servers to 1.4.19.1 last evening, but ended up having to drop back because of IAX calls failing at a near 50 % rate. Here is the message that we would receive on the console (multiple times), and then it would hangup the call. Avoiding IAX destroy deadlock Anyone else having similar problems? Thanks, Mike Clark ___ -- 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] Best Click-to-call client
I use 1ezphone because its not activex and works all operating systems and browser.Plus the codec is great and only uses 10k - Original Message - From: Steven To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Click-to-call client Date: Thu, 24 Apr 2008 08:20:15 -0400 I use click2call. http://www.geocities.com/babarnazmi/index2.htm (it is really a click to talk, as I removed the dialing capabilities and hardcoded the extension) It is an activex control though. All of my testing has shown it be be pretty clean. We have it on our "contact us" page of our website and we also give that url to overseas (India, Germany, Japan) contacts and some have used it. Some do not want to open up the iax2 port in their firewall, but that is their issue. I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was easier for people to use than all of the RTP ports required for SIP. -- -- Steven http://teamvie.blogspot.com/ http://www.connectech.org/ "equis software" wrote in message news:[EMAIL PROTECTED] Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- 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 -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.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
[asterisk-users] ring group question
Hi All,
I'm trying to configure a ringgroup, which will ring the extension in the
group one by one. this is what i tried on my extension.conf
[macro-dial-ringgroup]
exten => s,1,Dial(SIP/${ARG1},15)
exten => s,n,NoOp( Dial Status: ${DIALSTATUS})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-CHANUNAVAIL,1,SetCallerId(${CALLERIDNUM})
exten => s-CHANUNAVAIL,n,Dial(SIP/${ARG1},15)
exten => s-BUSY,1,SetCallerId(${CALLERIDNUM})
exten => s-BUSY,n,Dial(SIP/${ARG1},15)
exten => s-NOANSWER,1,SetCallerId(${CALLERIDNUM})
exten => s-NOANSWER,n,Dial(SIP/${ARG1},15)
[ringgroup-1]
exten => 5000,1,Macro(dial-ringgroup,1100)
exten => 5000,n,Macro(dial-ringgroup,1101)
exten => 5000,n,Macro(dial-ringgroup,1102)
exten => 5000,n,Hangup
so when i dial 5000 it will ring 1100 no answer,or busy on 1100.
it will go to another extension which is 1101 and so on.
I have tried 5000,1,Dial(SIP/1100&SIP/1100) <--- this one works,
ringing at the same time, how can i do it in sequential?
hope anyone can help me. thank you
Ron
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Re: [asterisk-users] No DTMF on Sip Trunk?
On Thu, 2008-04-24 at 12:02 -0400, Noah Miller wrote: > For the first time, I'm setting up SIP trunking between two asterisk > boxes. The calls themselves work fine, but I'm not able to get DTMF > working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set "rfc2833compensate=yes" in the peer or friend section of sip.conf on the Asterisk 1.4 box. This tells Asterisk to send RFC2833 DTMF the way that Asterisk 1.2 expects it, instead of the newer (read: more standards compliant) way that Asterisk 1.4 now handles RFC2833 DTMF tones. In a nutshell, try adding "rfc2833compensate=yes" to your section named [129trunk551] on the box you're calling Asterisk2. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] Playing mp3-files – will it b e OK?
Hi Harry - > 99% of all my users are calling from GSM phones, and my system > basically just plays some sound files back. > > The PBX is connected to an ISDN-30 connection. Are there any modules > for playing MP3 files, so I can use them with commands like Play() and > Background()? See asterisk-addons for the mp3 module. > And will it have any effect on the quality? The callers should hear the file at the codec-quality of the channel they're connecting on. So for your ISDN callers, that's probably ulaw or alaw, and for the internal phones, GSM. - Noah ___ -- 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] Playing mp3-files – will it b e OK?
2008/4/24 Jared Smith <[EMAIL PROTECTED]>: > On Thu, 2008-04-24 at 17:50 +0200, harry wrote: > > The PBX is connected to an ISDN-30 connection. Are there any modules > > for playing MP3 files, so I can use them with commands like Play() and > > Background()? > > If I were you, I'd transcode the files to alaw and play back the alaw > version, so that Asterisk doesn't have to transcode them for every call > (which is a waste of CPU cycles). > > If you *really* want to use MP3 files, you'll need to load the > format_mp3 module from asterisk-addons package. > You will also need to take into account that the music will be GSM compressed by a GSM mobile network, and that gives reasonably poor results in must cases. Over ISDN, the results are excellent. Regards, 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
Re: [asterisk-users] Playing mp3-files – will it be OK?
On Thu, 2008-04-24 at 17:50 +0200, harry wrote: > The PBX is connected to an ISDN-30 connection. Are there any modules > for playing MP3 files, so I can use them with commands like Play() and > Background()? If I were you, I'd transcode the files to alaw and play back the alaw version, so that Asterisk doesn't have to transcode them for every call (which is a waste of CPU cycles). If you *really* want to use MP3 files, you'll need to load the format_mp3 module from asterisk-addons package. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] No DTMF on Sip Trunk?
Hi All - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. I've tried using inband, rfc2833 and auto, and none of them work. Maybe I'm missing something obvious? Here's my config: Asterisk1 (1.2.18): sip.conf [129trunk551] type=friend secret= username=129trunk551 host=xxx.xxx.xxx.xxx context=phones dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Asterisk2 (ABE revC): sip.conf [129trunk551] type=friend secret=*** username=129trunk551 host=yyy.yyy.yyy.yyy context=default dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Thanks, Noah ___ -- 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] Playing mp3-files – will it b e OK?
Hello 99% of all my users are calling from GSM phones, and my system basically just plays some sound files back. The PBX is connected to an ISDN-30 connection. Are there any modules for playing MP3 files, so I can use them with commands like Play() and Background()? And will it have any effect on the quality? Load issues should be a problem, the number of concurrent calls are pretty low. Thanks ___ -- 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] Invitation to connect on LinkedIn
Dear Brian, On Thu, Apr 24, 2008 at 08:23:29AM -0700, Brian Nehring wrote: > > I was playing around and found some option to cross-reference all > gmail contacts and linkedin people. It's a weird, enlightening list, > so I figured I'd check the boxes of people I actually might know > (i.e., not random HR people, website admins, tech support, etc). Aparantly you Asterisk all too well. It's good to know it's not a random website admin. > Accept Brian Nehring's invite: > https://www.linkedin.com/e/isd/254835080/bC5K22oh/ Yeah, we're all your friends :-) (Ex-LinkedIn user Tzafrir) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Invitation to connect on LinkedIn
LinkedIn Asterisk, I was playing around and found some option to cross-reference all gmail contacts and linkedin people. It's a weird, enlightening list, so I figured I'd check the boxes of people I actually might know (i.e., not random HR people, website admins, tech support, etc). Cheers, -Brian Accept Brian Nehring's invite: https://www.linkedin.com/e/isd/254835080/bC5K22oh/ -- (c) 2008, LinkedIn Corporation ___ -- 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] Quality problems with ISDN PRI
Every CPU core shows up as a separate CPU under Linux. For those that have hyperthreaded processors, a single core processor will show up as two processors - assuming you have hyperthreading enabled. linuxian iandsd wrote: "top" says asterisk 1.2.25 is using multiple cores: Cpu0 : 2.7% us, 9.3% sy, 0.0% ni, 87.7% id, 0.0% wa, 0.3% hi, 0.0% si Cpu1 : 1.7% us, 4.0% sy, 0.0% ni, 94.3% id, 0.0% wa, 0.0% hi, 0.0% si Cpu2 : 1.3% us, 4.3% sy, 0.0% ni, 94.3% id, 0.0% wa, 0.0% hi, 0.0% si Cpu3 : 1.3% us, 3.0% sy, 0.0% ni, 95.6% id, 0.0% wa, 0.0% hi, 0.0% si is this multi-core ? I think its a multi-processor machine, and as i said I might be wrong simply because this bypasses by far my technical knowldge .. I m not a kernel developer after all. :) !DSPAM:4810121c213011316913527! ___ -- 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 !DSPAM:4810121c213011316913527! ___ -- 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] T.38 VoIP providers
Gafachi is the only one we have had success with for T38 fax. Jeff Johnson NeturallySpeaking Enterprise VoIP solutions at Small Business Prices (866) 448-0038 ext 103 (813) 774-3570 direct (813) 655-9049 fax www.neturallyspeaking.com http://www.neturallyspeaking.com/> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Thursday, April 24, 2008 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 VoIP providers By your experience, please someone tell me which T.38 capable VoIP SIP providers have you tested with success sending and receiving FAX with Asterisk 1.4. Thanks, Ricardo Carvalho. This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of the company. Warning: Although precautions have been taken to make sure no viruses are present in this email, the company cannot accept responsibility for any loss or damage that arise from the use of this email or attachments.___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Hello Ruben, Yes, if you consider using R2 support in chan_zap Unicall is no longer required. I will be not available online this weekend, please let me know your feedback after your try it. We can also meet via MSN so I can assist you in testing the next weeked (3-4 May). Thanks for the help. Moisés Silva On Thu, Apr 24, 2008 at 8:26 AM, Ruben Zamora <[EMAIL PROTECTED]> wrote: > Moises > > Thats means, that we arent going to use unicall? > > If that true i can test these weekend with a E1-Axtel. > > Thanks > > Ruben > > > Moises Silva escribió: > > > If you are an MFC/R2 user and want to help in the development of > > chan_zap support for this signalling, please take a look at the > > bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact > > me. Currently just México support is built-in, if you want your > > country variant supported, drop me a line. > > > > Moisés Silva > > > > > > > > ___ > -- 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 > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
> Hello Moisés, thanks for your effort on this! I would love to use Digium > cards for MFC/R2 signalling in the future. Currently you can use Digium cards with Unicall :-) , tho, having MFC/R2 on chan_zap is more handy. > I added some info you might like in the bugtracker, you might take a look at > it. I will, thanks! Moisés Silva -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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] Macro/Goto Help
Nevermind, helps when you reload the diaplan at BOTH ends :)
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, April 24, 2008 9:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Macro/Goto Help
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic from extensions.conf is below, first is from the system making the call,
the second is from the system receiving the call:
(CALLING SYSTEM)
The DUNDi system makes calls via IAX using a peer named priv
[local-dundi]
exten => _817NXX,1,Macro(dundi-lookup,${EXTEN})
exten => _817NXX,n,Macro(trunkdial,Zap/G0/w${EXTEN})
exten => _NXXNXX,1,Macro(trunkdial,Zap/G0/w${EXTEN})
[macro-dundi-lookup]
exten => s,1,Goto(${ARG1},1)
exten => s,n,MacroExit
include => dundi-priv-local
include => dundi-priv-lookup
include => dundi-e164-lookup
[dundi-priv-local]
exten => _4XX,1,Noop
[dundi-priv-lookup]
switch => DUNDi/priv
[dundi-e164-lookup]
switch => DUNDi/e164
(CALLED SYSTEM)
The IAX peer priv is dropped into the following context in the dialplan
[dundi-e164]
exten => _817.,1,Set(DID=${EXTEN:6})
exten => _817.,n,Noop(${DID})
exten => _817.,n,Set(GROUP(IAX)=incoming)
exten => _817.,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}>10]?fail)
exten => _817.,n,Goto(from-pri,${DID},1)
exten => _817.,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
If the total for all IAX calls is above 10, I want the call to fail so it'll
fall back and use ZAP instead of IAX. Instead the call just hangs up at the
CALLING system.
The from-pri logic has been excluded since it has no bearing on the question at
hand.
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This e-mail, facsimile, or letter and any files or attachments transmitted with
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any disclosure, copying, printing, or use of this information is strictly
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[asterisk-users] Macro/Goto Help
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic from extensions.conf is below, first is from the system making the call,
the second is from the system receiving the call:
(CALLING SYSTEM)
The DUNDi system makes calls via IAX using a peer named priv
[local-dundi]
exten => _817NXX,1,Macro(dundi-lookup,${EXTEN})
exten => _817NXX,n,Macro(trunkdial,Zap/G0/w${EXTEN})
exten => _NXXNXX,1,Macro(trunkdial,Zap/G0/w${EXTEN})
[macro-dundi-lookup]
exten => s,1,Goto(${ARG1},1)
exten => s,n,MacroExit
include => dundi-priv-local
include => dundi-priv-lookup
include => dundi-e164-lookup
[dundi-priv-local]
exten => _4XX,1,Noop
[dundi-priv-lookup]
switch => DUNDi/priv
[dundi-e164-lookup]
switch => DUNDi/e164
(CALLED SYSTEM)
The IAX peer priv is dropped into the following context in the dialplan
[dundi-e164]
exten => _817.,1,Set(DID=${EXTEN:6})
exten => _817.,n,Noop(${DID})
exten => _817.,n,Set(GROUP(IAX)=incoming)
exten => _817.,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}>10]?fail)
exten => _817.,n,Goto(from-pri,${DID},1)
exten => _817.,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
If the total for all IAX calls is above 10, I want the call to fail so it'll
fall back and use ZAP instead of IAX. Instead the call just hangs up at the
CALLING system.
The from-pri logic has been excluded since it has no bearing on the question at
hand.
This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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Re: [asterisk-users] AST-2008-006 - 3-way handshake in IAX2 incomplete
On Thu, 2008-04-24 at 09:13 -0500, Tilghman Lesher wrote: > > Check the archives. Indeed, you are correct. My apologies. I forgot that I temporarily unsubbed from the -users list for a period of time where I was just getting too much volume of e-mail and asterisk-users had to be one of the ones to go. I've just subbed to the -security list so that if I again need to suspend -users I won't miss the advisories. > In short, I can't think of a reason why you should be unaware of any security > advisory regarding a past release of Asterisk. Other than the above "volume management" problem at my end. :-) Cheers, b. signature.asc Description: This is a digitally signed message part ___ -- 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] G723 pass thru
More importantly, for it to "pass-through" you need something that processes g723 on the other end. If Asterisk is terminating the call by handing it off to the PSTN or to another phone that does not do g723 then Asterisk must transcode and that requires the license. Eric Wieling wrote: > allow=g723.1 or allow=g723 (I don't remember which). > > aby azid wrote: > >> Hi, >> >> I have softphone with a g723 codec, my question is how do i set it as Pass >> thru in Asterisk? >> > > > -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] Next step in extensions.conf after answer the phone in Queue
In article <[EMAIL PROTECTED]>, Atis Lezdins <[EMAIL PROTECTED]> wrote: > > Atis Lezdins wrote: > > > Queue will continue if called person hangs up (and there's no option). > > > If caller hangs up, call goes to h extension in same context. Just the > > > same way as Dial with 'g'. There's a change in 1.6 that allows called > > > channel to continue if caller hangs up, so probably something like > > > this could be applied also to Queue (or was that actually working with > > > using Local channels?). > > > > > On Wed, Apr 23, 2008 at 8:18 PM, Al Baker <[EMAIL PROTECTED]> wrote: > > Why would you want a "channel to continue" after the caller has hung up. > > I clearly am missing something here because I can't see what good that > > would be. What do people do with this "Continued Channel" ? > > What is is used for ? How Does having it help you ? ??? > > To play something to called party. > > I'm not familiar with that feature too deep, but I guess it's not > caller channel but called channel that's continued. No. The dialplan is executing on the calling channel. The called channel just belongs to the Dial application and is not in the dialplan itself. So the called channel has no context in which to invoke a Playback() when the caller hangs up. This has recently been addressed in SVN trunk by the addition of the option F(context^exten^pri) - When the caller hangs up, transfer the called party to the specified context and extension and continue execution. However, it doesn't appear to be in the 1.6.0 branch, so won't appear in a release until 1.6.1. If you want to apply the patch yourself, you can find it in the bug tracker at http://bugs.digium.com/view.php?id=11954 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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] Next step in extensions.conf after answer the phone in Queue
Atis Lezdins wrote: >> Atis Lezdins wrote: >> > Queue will continue if called person hangs up (and there's no option). >> > If caller hangs up, call goes to h extension in same context. Just the >> > same way as Dial with 'g'. There's a change in 1.6 that allows called >> > channel to continue if caller hangs up, so probably something like >> > this could be applied also to Queue (or was that actually working with >> > using Local channels?). >> > >> > > On Wed, Apr 23, 2008 at 8:18 PM, Al Baker <[EMAIL PROTECTED]> wrote: > >> Why would you want a "channel to continue" after the caller has hung up. >> I clearly am missing something here because I can't see what good that >> would be. What do people do with this "Continued Channel" ? >> What is is used for ? How Does having it help you ? ??? >> > > To play something to called party. > > I'm not familiar with that feature too deep, but I guess it's not > caller channel but called channel that's continued. > > Regards, > Atis > > > I am guessing something to the tune of " missed a call from press 1 to call them back now.". That is a good feature idea. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- 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] Forking in Dialplan
On Thursday 24 April 2008 03:51, Tobias Ahlander wrote: > Is it possible to somehow fork in the dialplan? Say a call comes in. Then I > want to wait 30 seconds and then write in a database, but at the same time > while I wait I want to go on with other commands too. There isn't a fork, but there is a method built in that is rather similar to the alarm(2) interface in the kernel, which is called Absolute Timeout. The method of using this is Set(TIMEOUT(absolute)=30), which sets a timer that will fire in 30 seconds. When that timer fires, any application that is currently executing will terminate, and you will be redirected to the "T" extension in the current context. Note that you can also cancel this timer by using setting the timeout to 0, i.e. Set(TIMEOUT(absolute)=0). Also note that there can only be a single absolute timeout per channel (i.e. setting a new timeout resets any timer currently in effect). -- Tilghman ___ -- 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] AST-2008-006 - 3-way handshake in IA X2 incomplete
On Wednesday 23 April 2008 18:26, Brian J. Murrell wrote: > On Wed, 2008-04-23 at 08:52 -0500, Tilghman Lesher wrote: > > Please understand that that's NOT the only security fix that has gone in > > during that time. If this is the only thing that you fix, you're likely > > to be vulnerable on several other levels. See our full list of security > > disclosures at http://downloads.digium.com/pub/security/ > > Hrm. Interesting. I don't recall seeing any of those others, such as > AST-2008-005 on this list. Is there some kind of "threat level" > threshold that's applied to what makes the list(s) and what doesn't? Check the archives. Every single one of the advisories goes out to -users, -dev, -announce, and -security, along with 4 outside lists (bugtraq, voipsec, full disclosure, and one other that I can't think of at the moment). The advisories are also posted at asterisk.org, and I think most of the people who blog on Asterisk pick up the advisories, as well. In short, I can't think of a reason why you should be unaware of any security advisory regarding a past release of Asterisk. -- Tilghman ___ -- 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] Next step in extensions.conf after answer the phone in Queue
> Atis Lezdins wrote: > > Queue will continue if called person hangs up (and there's no option). > > If caller hangs up, call goes to h extension in same context. Just the > > same way as Dial with 'g'. There's a change in 1.6 that allows called > > channel to continue if caller hangs up, so probably something like > > this could be applied also to Queue (or was that actually working with > > using Local channels?). > > On Wed, Apr 23, 2008 at 8:18 PM, Al Baker <[EMAIL PROTECTED]> wrote: > Why would you want a "channel to continue" after the caller has hung up. > I clearly am missing something here because I can't see what good that > would be. What do people do with this "Continued Channel" ? > What is is used for ? How Does having it help you ? ??? To play something to called party. I'm not familiar with that feature too deep, but I guess it's not caller channel but called channel that's continued. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Forking in Dialplan
- "Tobias Ahlander" <[EMAIL PROTECTED]> escreveu: Is it possible to somehow fork in the dialplan? Say a call comes in. Then I want to wait 30 seconds and then write in a database, but at the same time while I wait I want to go on with other commands too. On Thu, 24 Apr 2008, Vin??cius Fontes wrote: You can call an AGI script that will call another script. That last one would wait 10 seconds and write in the database. The following example works for me: /var/lib/asterisk/agi-bin/agi-test.agi: #!/bin/bash nohup /root/helloworld.sh 1>/dev/null 2>/dev/null & exit 0 /root/helloworld.sh: #!/bin/bash sleep 10 echo "Hello world!" >> /root/helloworld.txt exit 0 Why do you need the first AGI? Would: exten = _x.,n,system(nohup /root/helloworld.sh 1>/dev/null 2>&1 &) suit your needs? Thanks in advance, Steve Edwards [EMAIL PROTECTED] 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] DUNDi and SIP
I think I'm going to go about this a different way, if it works I'll post my
solution.
Essentially I'm going to limit the calls by grouping(didn't know you could use
categories until I did the research) and math. Limiting our corporate office
to 10 IAX calls, both incoming and outgoing together, and denying the call if
it's above that(sending chanunavail or something similar).
I'll then run all dials through a macro, looking up dundi routes. If it fails
I'll fall back to zap.
Thanks for the help though.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 23, 2008 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Take a look at this setup, it does not use passwords on the sip peers
or the mappings in Dundi. As long as you inside your network this
maybe the way to go.
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
You could also look at the incominglimit and outgoinglimit on IAX peers
On Wed, Apr 23, 2008 at 4:51 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> I'm fairly sure SIP will never work unless I hard-code peers everywhere,
> which isn't going to happen. The only reason I want to use it is for the
> call-limit option.
>
> Looking at sip channels there is no option to pass the extension after the
> IP, it's always [EMAIL PROTECTED], or [EMAIL PROTECTED], not [EMAIL
> PROTECTED]/extension or [EMAIL PROTECTED]/extension
>
> Looks like IAX and ZAP are the only two channel types that do a /extension
> type setup.
>
> Extensions.conf:
>
> [macro-dundi-lookup]
> exten => s,1,Goto(${ARG1},1)
> include => dundi-priv-local
> include => dundi-priv-lookup
>
> [dundi-priv-local]
> include => internal
>
> [dundi-priv-lookup]
> switch => DUNDi/priv
>
> Dundi.conf:
>
> [mappings]
> priv => dundi-priv-local,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
>
>
> Sent: Wednesday, April 23, 2008 4:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DUNDi and SIP
>
> Jeremy,
>
> It is not the dip peer that is failing but the dial plan:
>
>-- Goto (macro-dundi-lookup,400,1)
> [Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
> host: 192.168.4.51/400
> [Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full:
> Unable to create channel of type 'SIP' (cause 3 - No route to
> destination)
> == Everyone is busy/congested at this time (1:0/0/1)
>
> What is in the context macro-dundi-lookup?
>
> On Wed, Apr 23, 2008 at 12:47 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> > Nope..
> >
> > asterisk*CLI> dundi lookup [EMAIL PROTECTED]
> > 1. 0 SIP/priv:[EMAIL PROTECTED]/400 (EXISTS)
> > from 00:1e:0b:dd:e9:99, expires in 5 s
> > DUNDi lookup completed in 104 ms
> > -- Executing [EMAIL PROTECTED]:1] Set("SIP/156-08274b60",
> "CDR(accountcode)=wth") in new stack
> > -- Executing [EMAIL PROTECTED]:2] Set("SIP/156-08274b60",
> "CALLERID(all)=Corporate <100>") in new stack
> > -- Executing [EMAIL PROTECTED]:3] Macro("SIP/156-08274b60",
> "dundi-lookup|400") in new stack
> > -- Executing [EMAIL PROTECTED]:1] Goto("SIP/156-08274b60", "400|1") in
> new stack
> > -- Goto (macro-dundi-lookup,400,1)
> > [Apr 23 12:46:44] WARNING[2269]: chan_sip.c:2898 create_addr: No such
> host: 192.168.4.51/400
> > [Apr 23 12:46:44] WARNING[2269]: app_dial.c:1191 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination)
> > == Everyone is busy/congested at this time (1:0/0/1)
> > -- Executing [EMAIL PROTECTED]:4] Hangup("SIP/156-08274b60", "") in
> new stack
> > == Spawn extension (from-sip, 400, 4) exited non-zero on
> 'SIP/156-08274b60'
> >
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
> Reeves
> >
> >
> > Sent: Tuesday, April 22, 2008 10:36 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] DUNDi and SIP
> >
> > Try this,
> >
> > [priv]
> > dbsecret=dundi/secret
> > disallow=all
> > allow=ulaw
> > canreinvite=no
> > nat=no
> > context=from-internal
> > type=friend
> >
> > priv => dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
> >
> >
> >
> > On Tue, Apr 22, 2008 at 8:23 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote:
> > > No.
> > >
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce
> Reeves
> > >
> > >
> > > Sent: Tuesday, April 22, 2008 6:00 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] DUNDi and SIP
> > >
> > > Jeremy,
[asterisk-users] T.38 VoIP providers
By your experience, please someone tell me which T.38 capable VoIP SIP providers have you tested with success sending and receiving FAX with Asterisk 1.4. Thanks, Ricardo Carvalho. ___ -- 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] G723 pass thru
allow=g723.1 or allow=g723 (I don't remember which). aby azid wrote: > Hi, > > I have softphone with a g723 codec, my question is how do i set it as Pass > thru in Asterisk? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Moises Thats means, that we arent going to use unicall? If that true i can test these weekend with a E1-Axtel. Thanks Ruben Moises Silva escribió: > If you are an MFC/R2 user and want to help in the development of > chan_zap support for this signalling, please take a look at the > bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact > me. Currently just México support is built-in, if you want your > country variant supported, drop me a line. > > Moisés Silva > > ___ -- 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] Forking in Dialplan
You can call an AGI script that will call another script. That last one would wait 10 seconds and write in the database. The following example works for me: /var/lib/asterisk/agi-bin/agi-test.agi: #!/bin/bash nohup /root/helloworld.sh 1>/dev/null 2>/dev/null & exit 0 /root/helloworld.sh: #!/bin/bash sleep 10 echo "Hello world!" >> /root/helloworld.txt exit 0 Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Tobias Ahlander" <[EMAIL PROTECTED]> escreveu: > Hello, > > Is it possible to somehow fork in the dialplan? Say a call comes in. > Then I want to wait 30 seconds and then write in a database, but at > the same time while I wait I want to go on with other commands too. > > > Thanks, > Best regards, > Tobias > > ___ > -- 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] Best Click-to-call client
Do you have an example of it working on your website? When I try the click2call websitenone of the demo's actually work? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steven > Sent: Thursday, 24 April 2008 8:20 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Best Click-to-call client > > I use click2call. http://www.geocities.com/babarnazmi/index2.htm (it is > really a click > to talk, as I removed the dialing > capabilities and hardcoded the extension) > > It is an activex control though. > > All of my testing has shown it be be pretty clean. > > We have it on our "contact us" page of our website and we also give that url > to > overseas (India, Germany, Japan) contacts and some > have used it. > Some do not want to open up the iax2 port in their firewall, but that is > their issue. > > I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was > easier for > people to use than all of the RTP ports > required for SIP. > > -- > -- > Steven > > http://teamvie.blogspot.com/ > http://www.connectech.org/ > > > > "equis software" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > Hi, I need to make Click-to-Call web application to connect with an asterisk > server. > I´m using Java > What solution recommend me? > > Thanks > > > > > ___ > -- 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] prepaid on the trunks
You should also look at Darren's ASTPP, I am not sure if you missed
that earlier in the thread. It is basically ASTCC with major
improvements. It even has the ability to tie into OSCommerce which in
turn can connect to several credit card merchant accounts.
It is much more robust than ASTCC.
Thanks,
Steve Totaro
On Thu, Apr 24, 2008 at 12:17 AM, Nhadie Ramos <[EMAIL PROTECTED]> wrote:
> thank you sir, i will try to check on that. i haven't really tried astcc yet
> so i really dont understand how it works right now.
>
> also, do you have any reference on using app_prepaid? can't find some sample
> config, i would like to see how i can use that. do you think app_prepaid is
> suited for the scenario i have?
>
> thank you
>
> regards
> nhadie
>
>
>
> "Brian J. Murrell" <[EMAIL PROTECTED]> wrote:
>
> On Wed, 2008-04-23 at 15:41 -0600, Darren Wiebe wrote:
> > Ok, I'm not aware of this feature in astcc
>
> Keep in mind that astcc is simply a tool that keeps a database of
> minutes used for some "entity" (typically a calling card) and calculates
> those minutes used against a pre-charged amount. The "number" of the
> entity can be passed to astcc (i.e. so that it does not need to prompt
> the user for it) in such a way:
>
> exten => _1NXXNXX,n,DeadAGI(astcc.agi,${cardnum},${EXTEN})
>
> So binding a trunk to a "cardnum" (i.e. a given pre-charged account)
> should be easy enough to do.
>
> b.
>
> ___
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>
>
>
>
> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
> now.
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Re: [asterisk-users] Quality problems with ISDN PRI
> > There are much better solutions than doing a RAM drive. While it may > be stable (not in my experience, I advise using different servers for > different tasks (with redundancy obviously). A phone switch should be > just that, a recording server should also be just that (in demanding > environments). hi, still hoping you will give us some insight about remote recording server. ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Hello Moisés, thanks for your effort on this! I would love to use Digium cards for MFC/R2 signalling in the future. I added some info you might like in the bugtracker, you might take a look at it. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - "Moises Silva" <[EMAIL PROTECTED]> escreveu: > If you are an MFC/R2 user and want to help in the development of > chan_zap support for this signalling, please take a look at the > bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact > me. Currently just México support is built-in, if you want your > country variant supported, drop me a line. > > Moisés Silva > > -- > "I do not agree with what you have to say, but I'll defend to the > death your right to say it." Voltaire > > ___ > -- 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] Best Click-to-call client
I use click2call. http://www.geocities.com/babarnazmi/index2.htm (it is really a click to talk, as I removed the dialing capabilities and hardcoded the extension) It is an activex control though. All of my testing has shown it be be pretty clean. We have it on our "contact us" page of our website and we also give that url to overseas (India, Germany, Japan) contacts and some have used it. Some do not want to open up the iax2 port in their firewall, but that is their issue. I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was easier for people to use than all of the RTP ports required for SIP. -- -- Steven http://teamvie.blogspot.com/ http://www.connectech.org/ "equis software" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? Thanks ___ -- 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] Click-to-talk (Java application)
I use click2call. http://www.geocities.com/babarnazmi/index2.htm It is an activex control though. All of my testing has shown it be be pretty clean. We have it on our "contact us" page of our website and we also give that url to overseas (India, Germany, Japan) contacts and some have used it. Some do not want to open up the iax2 port in their firewall, but that is their issue. I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was easier for people to use than all of the RTP ports required for SIP. -- -- Steven http://teamvie.blogspot.com/ http://www.connectech.org/ "equis software" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] Hi! I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? ___ -- 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] Forking in Dialplan
what kind of command do you want it to do in the background? The obvious answer your question would probably be to use an agi script. On Thu, Apr 24, 2008 at 11:51 AM, Tobias Ahlander <[EMAIL PROTECTED]> wrote: > Hello, > > Is it possible to somehow fork in the dialplan? Say a call comes in. Then > I want to wait 30 seconds and then write in a database, but at the same time > while I wait I want to go on with other commands too. > > > Thanks, > Best regards, > Tobias > > ___ > -- 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 > -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- 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] G723 pass thru
Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid ___ -- 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] Forking in Dialplan
Hello, Is it possible to somehow fork in the dialplan? Say a call comes in. Then I want to wait 30 seconds and then write in a database, but at the same time while I wait I want to go on with other commands too. Thanks, Best regards, Tobias ___ -- 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] Newbie Polycom: Instant Messaging
Just want to know if anyone has used instant messaging using Polycom and Asterisk. >From Google, I did not really see IM being mentioned at all. It appears no one is interested to implement it in Asterisk. Or I guess people would rather use Jabber or other IM messengers. ___ -- 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] Gentilini, Paul is out of the office.
I will be out of the office starting 04/23/2008 and will not return until 04/29/2008. ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
If you are an MFC/R2 user and want to help in the development of chan_zap support for this signalling, please take a look at the bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact me. Currently just México support is built-in, if you want your country variant supported, drop me a line. Moisés Silva -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire ___ -- 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
