Re: [asterisk-users] Asterisk as an IVR solution
SO does that mean that if he used BACKGROUND is a SubRoutine he would get the correct or desired action , from his point of view? It would jump to the 1 Extension in the SUBROUTINE ? Tilghman Lesher wrote: On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 10, 2008 6:07:36 PM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Fri, Jul 11, 2008 at 8:28 AM, Douglas Garstang [EMAIL PROTECTED] wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. I do the IVR stuff in a context and jump to it as needed. The context is reusable from anywhere. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Analog lines dtmf problem
Hi
I have a problem with dtmf recognition an analog lines connected to Sangoma
A200. The digits (in most cases the first one) are doubled and so my IVR is
useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but
nothing worked. I also noticed one thing it only happens during the
background application:
exten = s,1,Background(soundfile)
exten = 111,1,Dial(SIP/111)
exten = 122,1,Dial(SIP/122)
it never happens in WaitExten application:
exten = s,1,WaitExten(25)
exten = _XX.,1,Noop(${EXTEN}) in ${EXTEN} I always have all digits
that I pressed in correct order and amount.
Can you tell me what cause this:
is it the background application (the bug in asterisk), analog lines (in all
my installations of asterisk where I utilize the A200 card I have some
problems with dtmf), or the card (maybe digium cards are better in DFTM
recognition).
I asked sangoma tech support for help but so far with no result.
I have preform some tests and I have connected the two asterisk PBX one with
digium TDM400 and fxs ports with another asterisk with A200 sangoma card and
fxo ports and in this case dtmf recognition is 100% accurate. The same
happens when I connect the SIP ATA with fxs port to sangoma fxo port - 100%
OK.
What do you think about this?
Cheers
Andrew
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[asterisk-users] Microsoft CRM 4.0 integration with asterisk
Hello ! I am wondering if anyone has experiences with the integration of Asterisk 1.4.19 into Microsoft Dynamics CRM 4.0 ? Or alternatively integration with Microsoft Office Communications server (however trying to avoid this, if it isn't really necessary for the integration). I would be glad to receive any links or manuals on this topic, which helped you to integrate it. Kind regards, Jan Prunk -- Jan Prunk janprunk AT SPAMFREE gmail DOT com Website: http://www.prunk.si PGP key: 00E80E86 Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing inbuilt sound messages
I was curious so I took a look at my sounds directory. Most of the
files are 644 except the g729 which are 444. I also noticed that the
ownerid/groupid are a non-existent 1000/1000. I take it that the sound
installer uses something like tar with the option to keep the original
owner and groupid which it shouldn't be doing. If it's tar it should
use at least the option -o when doing the extraction to
/var/lib/asterisk/sounds.
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.alaw
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.g722
-r--r--r-- 1 1000 1000 870 Dec 5 2007 zip-code.g729
-rw-r--r-- 1 1000 10001452 Dec 5 2007 zip-code.gsm
-rw-r--r-- 1 1000 10006985 Dec 5 2007 zip-code.ulaw
-rw-r--r-- 1 1000 1000 14014 Dec 5 2007 zip-code.wav
from asterisk/sounds/Makefile:
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR); cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/es; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/fr; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR); cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/es; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst
$(SOUNDS_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(SOUNDS_DIR)/fr; cat $(PWD)/$${PACKAGE} | gzip -d |
tar xf -) \
Makefile: @PACKAGE=$(subst $(MOH_DIR)/.asterisk,asterisk,$@).tar.gz; \
Makefile: (cd $(MOH_DIR); cat $(PWD)/$${PACKAGE} | gzip -d | tar
xf -) \
Tzafrir Cohen wrote:
On Fri, Jul 11, 2008 at 09:56:29AM +1200, Lists wrote:
I only did the 420 because thats what the original files looked like?
r-- -w- ---
Should I change this to 644?
Yes!
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Re: [asterisk-users] Tracking Call Time While in Dial()
Call an AGI right before the start of the Dial command to record the start time and ether use an manager application (makes use of manager API) or call an DeadAGI once the call has ended (from the h extension). This requires a bit of programming - but then again some programming is required anyway to display the actual talk time somewhere. It might also be that I'm an programmer and I attempt to solve all problems writing programs, so maybe someone else has a better idea! -- Cosmin Prund De la: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Douglas Garstang Trimis: Thursday, July 10, 2008 7:49 PM Către: asterisk-users@lists.digium.com Subiect: [asterisk-users] Tracking Call Time While in Dial() So, I've been asked if this is possible. Someone wants to actively monitor the duration of a call, while the call is still in progress. Obviously, in Asterisk, once the Dial() application starts, you lose dial plan control until after the call has ended, successful or otherwise. Anyone know if that kind of thing is possible? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C450 broken rtp handling
Hello, I've got a problem with rtp handling by siemens c450 and similar. I experience a couple seconds of silence between early media and normal call (normal call's rtp is dropped by phone). This is caused by SSRC changing (even though marker bit is set). I have all relevant patches applied - it still happens on 1.4.21.1 and every version before that. Especially http://bugs.digium.com/view.php?id=12570 doesn't change anything, because call is p2p bridged. Issue can be fixed by forcing use of the same ssrc in ast_raw_write and bridged rtp writes and my custom patch works, but I don't want to use it if it can be done in some other way. Is there a way to force treating outgoing rtp as one stream, instead of switching source after early media? Is there a way to do it without resigning from p2p bridging? Thanks for ideas, Stan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Call Screener
On Thu, 2008-07-10 at 10:38 -0500, Jared Smith wrote: On Wed, 2008-07-09 at 17:54 -0400, Ryan M. Colbert wrote: I'm trying to build a simple accept/reject screening app for inbound calls that * forwards to my cell phone. Basically I want * to announce the caller ID and then let me press 1 to accept the call or 2 to reject the call and send the outside party to voicemail. While you can certainly do it by using a dial macro, a simpler method is to check out the p and P options to the Dial() application. Yes, a couple years back, I added quite a bit of code to Dial() to support several forms of call screening. See the docs for Dial (core show application dial) for the P and p options, with modifiers N and n. You can use the astdb features to remember your choices for callers based on cid and record and store announcements that you can use from the dialplan to announce calls based on callerid. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming
Folks: This is my first post, so please let me know if I transgress in any way... In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. Many thanks, --ag -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
Thanks, but that won't do what I need. By calling an AGI before the call starts and after the call ends, all I am doing is accounting the start and the end of the call, not actively monitoring the duration of the call as it occurs. - Original Message From: Cosmin Prund [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 3:57:23 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() Call an AGI right before the start of the Dial command to record the start time and ether use an manager application (makes use of manager API) or call an DeadAGI once the call has ended (from the h extension). This requires a bit of programming - but then again some programming is required anyway to display the actual talk time somewhere. It might also be that I'm an programmer and I attempt to solve all problems writing programs, so maybe someone else has a better idea! -- Cosmin Prund De la:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] În numele Douglas Garstang Trimis: Thursday, July 10, 2008 7:49 PM Către: asterisk-users@lists.digium.com Subiect: [asterisk-users] Tracking Call Time While in Dial() So, I've been asked if this is possible. Someone wants to actively monitor the duration of a call, while the call is still in progress. Obviously, in Asterisk, once the Dial() application starts, you lose dial plan control until after the call has ended, successful or otherwise. Anyone know if that kind of thing is possible? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 01:28:34 Douglas Garstang wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. That's the point. A Macro is NOT a subroutine. It's like saying that you can't effectively hammer a nail with a screwdriver, and therefore you think the screwdriver has a known problem. There's nothing wrong with the screwdriver; it simply is the wrong tool for the job. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Yes, and by doing that your compounding the fact that all your variables are global. - Original Message From: randulo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 12:14:28 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Fri, Jul 11, 2008 at 8:28 AM, Douglas Garstang [EMAIL PROTECTED] wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. I do the IVR stuff in a context and jump to it as needed. The context is reusable from anywhere. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 01:05:22 Al Baker wrote: Tilghman Lesher wrote: On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. SO does that mean that if he used BACKGROUND is a SubRoutine he would get the correct or desired action , from his point of view? It would jump to the 1 Extension in the SUBROUTINE ? Yes, if he used Background within a Gosub, it would behave the way that he expects. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Microsoft CRM 4.0 integration with asterisk
I dont know if this will help, but I have been working with MS OCS at work, and * 1.6 integrates rather wall tith OCS speech server. If you need help on that relm, I can try to help. (admitidly I dont have inbound calls working, but we arnt worried about that, as our appplication is strictly outbound.) --Chris On Fri, Jul 11, 2008 at 2:16 AM, Jan Prunk [EMAIL PROTECTED] wrote: Hello ! I am wondering if anyone has experiences with the integration of Asterisk 1.4.19 into Microsoft Dynamics CRM 4.0 ? Or alternatively integration with Microsoft Office Communications server (however trying to avoid this, if it isn't really necessary for the integration). I would be glad to receive any links or manuals on this topic, which helped you to integrate it. Kind regards, Jan Prunk -- Jan Prunk janprunk AT SPAMFREE gmail DOT com Website: http://www.prunk.si PGP key: 00E80E86 Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- find / -name *base* -user your -print | xargs 'chown us' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 09:22:25 Douglas Garstang wrote: Yes, and by doing that your compounding the fact that all your variables are global. No, his variables are local to the channel he's using. Global variables are a completely different beast. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
On Friday 11 July 2008 09:21:56 Douglas Garstang wrote: Thanks, but that won't do what I need. By calling an AGI before the call starts and after the call ends, all I am doing is accounting the start and the end of the call, not actively monitoring the duration of the call as it occurs. It is unclear from your description what you want to do. Could you be more explicit? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:20:40 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 01:28:34 Douglas Garstang wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. That's the point. A Macro is NOT a subroutine. It's like saying that you can't effectively hammer a nail with a screwdriver, and therefore you think the screwdriver has a known problem. There's nothing wrong with the screwdriver; it simply is the wrong tool for the job. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming
On Fri, 11 Jul 2008, Artie Gold wrote: This is my first post, so please let me know if I transgress in any way... A more meaningful subject would get more interest. It also helps when someone else is searching the mailing list archives. For example, SIP timing out over satellite. In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP timing out over satellite connection on 1.4.21 (works with 1.4.18.1)
In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. Many thanks, --ag [This is an edited repost with an improved subject line -- thanks to Steve Edwards] -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Ugh. Yes, the variables are local to the current channel. However, they are global to the entire dial plan within the current channel. I have stepped on myself many times because I've had a loop counter called $i for example, jumped somewhere else within that loop, reused the same variable name, $i, and screwed up my logic. Surely you where aware that's the type of thing I was talking about. I'd be surprised if you didn't. Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:36:54 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 09:22:25 Douglas Garstang wrote: Yes, and by doing that your compounding the fact that all your variables are global. No, his variables are local to the channel he's using. Global variables are a completely different beast. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Fails to convert INFO to Inband
Hi, We are using asterisk 1.4.20 load. We have seen that couple of times Asterisk fails to convert SIP INFO packet in to Inband tone. Problem Description: Asterisk behaves as a media proxy between proxy1 and proxy2. Proxy1 transmit DTMF using SIP INFO method while proxy2 works with Inaband Tone. I have provided complete setup descriptions below. After analyzing the logs we found that when proxy1 transmit 1st DTMF Digit it successfully converts into inband but when we send next digit chan-generatordata is set to NULL therefore no Inabdn tone generate further. This remains set to NULL for a while and then it start converting again into inband. Here is sip.conf [general] Dtmfmode=inband [peer-proxy1] type=peer ; we only want to call out, not be called fromdomain=proxy2.varaha.com ; host= 71.153.215.73; context=from-proxy1 ;dtmfmode=inband dtmfmode=info rtpkeepalive=1 port=5045 disallow=all allow=ulaw allow=alaw [peer-proxy2] type=peer ; we only want to call out, not be called fromdomain= proxy1.varaha.com context=from-proxy2 ;dtmfmode=auto ;dtmfmode=rfc2833 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. ;dtmfmode=inband ;dtmfmode=info ;rtpkeepalive=1 port=5080 disallow=all allow=ulaw allow=alaw We have seen following logs in message file [Jul 11 03:58:22] DTMF[32393] chan_sip.c: -Got INFO with: [1] Duration: [250] [Jul 11 03:58:22] DTMF[32508] channel.c: DTMF end '1' received on SIP/trunk.myvtel.com-090f4880, duration 250 ms [Jul 11 03:58:22] DTMF[32508] channel.c: DTMF begin emulation of '1' with duration 250 queued on SIP/trunk.myvtel.com-090f4880 [Jul 11 03:58:22] NOTICE[32508] channel.c: Send DTMF in Inband: [1] [Jul 11 03:58:22] NOTICE[32508] channel.c: Result of ast_playtones_start: [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: Generate DTMF for pointer: [0x9103E20] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x9103E20] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] DTMF[32508] channel.c: DTMF end emulation of '1' queued on SIP/trunk.myvtel.com-090f4880 [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x0] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x0] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions: chan-generatordata[0x0] ast_internal_timing_enabled(chan): [0] [Jul 11 03:58:22] NOTICE[32508] channel.c: ast_read_generator_actions:
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound
but I cant hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
If Asterisk says it is playing the file, then I would suspect the file
itself has nothing to say. Try copying the file to your computer and
playing it. If it does indeed play locally on your computer with
audio, double check to make sure it is in the right format. I use AGI
to play files all the time. Actually, I use an AGI script as my whole
menu and dialing system to replace having to do it in AEL (so much
nicer to add a single MySQL record and suddenly have voicemail and
direct dial work instantly).
Daniel
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera
de lo comun
*---*
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Re: [asterisk-users] Asterisk as an IVR solution
On Fri, 11 Jul 2008, Douglas Garstang wrote: Ugh. Yes, the variables are local to the current channel. However, they are global to the entire dial plan within the current channel. I have stepped on myself many times because I've had a loop counter called $i for example, jumped somewhere else within that loop, reused the same variable name, $i, and screwed up my logic. Ugh indeed. While I sympathize with your local/global name space issues, you lose credibility with your false economy. IMNSHO, anybody who uses a single [common] letter for a variable deserves a bump in the temperature when they reach their final resting place :) For example, out of the 157 applications on one of my Asterisk servers, 76 contain the letter l. (absolutetimeout, adsiprog, agentcallbacklogin, agentlogin, agentmonitoroutgoing, agi, alarmreceiver, appendcdruserfield, authenticate, changemonitor, chanisavail, congestion, datetime, deadagi, dial, dictate, digittimeout, directory, disa, dundilookup, eagi, endwhile, execif, execiftime, externalivr, festival, getcpeid, gosubif, gotoif, gotoiftime, hasnewvoicemail, hasvoicemail, iax2provision, ices, importvar, lookupblacklist, lookupcidname, macroexit, macroif, mailboxexists, meetmeadmin, milliwatt, mixmonitor, monitor, pickup, privacymanager, readfile, realtime, realtimeupdate, responsetimeout, retrydial, ringing, saydigits, sayphonetic, sayunixtime, sendimage, setcallerid, setcdruserfield, setcidname, setcidnum, setrdnis, settransfercapabilit, sipaddheader, sipdtmfmode, sipgetheader, stopmonitor, testclient, txtcidname, vmauthenticate, voicemail, voicemailmain, wait, waitexten, waitforring, waitforsilence, while) Surely you can come up with a name slightly more descriptive -- maybe idx? Take pity on the next sod that has to plod through your dialplan. The millisecond you spend typing a more meaningful name will be returned to you (or your employer) a millionfold. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:36:54 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 09:22:25 Douglas Garstang wrote: Yes, and by doing that your compounding the fact that all your variables are global. No, his variables are local to the channel he's using. Global variables are a completely different beast. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 09:40:55 Douglas Garstang wrote: Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). I've mentioned Gosub at least twice before in this thread, which implements a subroutine. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing calls but no incoming calls with X100P
Hi all,
I have a problem with my asterisk box and an X100P FXO card. I am able to
place outgoing calls from my SIP phone (Cisco 7940) to any external number
using my PSTN line, but when I call my PSTN line from my cell phone, the
Cisco doesn't ring (and no message appears in the Asterisk CLI).
Here are my config files:
zaptel.conf
fxsks=1
loadzone = be
defaultzone = be
zapata.conf
[channels]
context=incoming_calls
usecallerid=yes
hidecallerid=no
immediate=no
signalling=fxs_ks
callerid = test 123
echocancel=yes
group=1
channel=1
extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
exten = s,1,Verbose(1|Unrouted call handler)
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Playback(tt-weasels)
exten = s,n,HangUp()
[incoming_calls]
;exten = _X.,1,NoOp()
;exten = _X.,n,Dial(SIP/1000)
;exten = _X.,1,Dial(SIP/1000)
exten = s,1,Dial(SIP/1000,20,tr)
[outgoing_calls]
exten = _X.,1,NoOp()
exten = _X.,n,Dial(Zap/1/${EXTEN})
[internal]
exten = 1000,1,Verbose(1|Extension 1000)
exten = 1000,n,Dial(SIP/1000,30)
exten = 1000,n,HangUp()
exten = 500,1,Verbose(1|Echo test application)
exten = 500,n,Echo()
exten = 500,n,HangUp()
[phones]
;include = internal
;include = incoming_calls
include = outgoing_calls
sip.conf
[general]
context=phones
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[1000]
type=friend
context=phones
host=dynamic
username=1000
secret=1000
allow=all
zttool shows that the card is working fine and is configured. When I use
ztcfg I get no errors. I've been looking for this for days now and it's
driving me nuts. I read somewhere that you may have to disable
AUDIO_RINGCHECK in the wctdm driver, but I'm using the wcfxo (listed in
lsmod together with zaptel).
All help is appreciated!
Thanks,
Tom
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Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming
On Friday 11 July 2008 09:17:37 Artie Gold wrote: In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. Try setting t1min to something higher than the default, 100 (ms). This value is settable globally, as well as per-peer. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing dropped calls...
The other thing that baffles me about this setup is that it only seems to happen to people who are connected to the internal network in the office. They have about 30 remote users that have not reported this same problem, their issue is usually bandwidth related from their home connection. We have checked the internal network several times and there is not any obvious problem (apart from the dropped calls). They use high end Cisco switches and they were just audited to make sure there were no configuration errors. All the internal phones are Aastra (most are 9133i and some others 53i). On Thu, 2008-07-10 at 20:55 -0400, Steve Totaro wrote: Try dropping the IAX2 and only use SIP. Don't ask why? Just give it a try and see if things improve for you. Also when you assume, you make and ass out of you and me (just a little joke, get it? ass-u-me.) You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly (point to point) with G729 or just get a real ISDN or POTS lines. And then you will still have dropped calls depending on your volume and how vocal your users are. Usually, once they perceive a problem, then even if the other side of the call is on a cell and the cell drops the call, you will get a complaint. The only way to track those down are on a case by case basis with ANI II codes 61-63 http://www.nanpa.com/number_resource_info/ani_ii_assignments.html Thanks, Steve Totaro On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez [EMAIL PROTECTED] wrote: My customer has a 10mpbs fiber connection to the Internet so we have always assumed that the connection is not really a problem. We will look into it. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
On 07/11/08 18:37, Dave Cotton wrote: SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC If you plug it into to UPS some of them have protection for phone lines, it should protect it from lightning. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
Dave Cotton wrote: SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC Surge protection is your friend. :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
Joseph wrote: On 07/11/08 18:37, Dave Cotton wrote: SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC If you plug it into to UPS some of them have protection for phone lines, it should protect it from lightning. Should is the operative word. They didn't. DC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming
This is a quite promising idea. Many thanks. I'll post my results to the list... Cheers, --ag On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 11 July 2008 09:17:37 Artie Gold wrote: In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. Try setting t1min to something higher than the default, 100 (ms). This value is settable globally, as well as per-peer. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Artie Gold F4W, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
I want to track call duration while the call is in progress. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:39:40 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Friday 11 July 2008 09:21:56 Douglas Garstang wrote: Thanks, but that won't do what I need. By calling an AGI before the call starts and after the call ends, all I am doing is accounting the start and the end of the call, not actively monitoring the duration of the call as it occurs. It is unclear from your description what you want to do. Could you be more explicit? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
A subroutine with arguments? - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 8:58:46 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 09:40:55 Douglas Garstang wrote: Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). I've mentioned Gosub at least twice before in this thread, which implements a subroutine. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement --- SPA3102 how reliable is it?
Dave Cotton wrote: Joseph wrote: On 07/11/08 18:37, Dave Cotton wrote: SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC If you plug it into to UPS some of them have protection for phone lines, it should protect it from lightning. Should is the operative word. They didn't. DC I'm very suspicious of the effectiveness of the things they put in low end UPSes. However, if you buy the kind of lightning suppressor that is attached to phone lines as the enter your house, and put one at each end of your 3km of cable, it should help a lot. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Fine, I'll call it ${LoopVariable} then... how's that going to fix the problem?
- Original Message
From: Steve Edwards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 8:43:47 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Ugh. Yes, the variables are local to the current channel. However, they
are global to the entire dial plan within the current channel. I have
stepped on myself many times because I've had a loop counter called $i
for example, jumped somewhere else within that loop, reused the same
variable name, $i, and screwed up my logic.
Ugh indeed. While I sympathize with your local/global name space issues,
you lose credibility with your false economy.
IMNSHO, anybody who uses a single [common] letter for a variable deserves
a bump in the temperature when they reach their final resting place :)
For example, out of the 157 applications on one of my Asterisk servers, 76
contain the letter l.
(absolutetimeout, adsiprog, agentcallbacklogin, agentlogin,
agentmonitoroutgoing, agi, alarmreceiver, appendcdruserfield,
authenticate, changemonitor, chanisavail, congestion, datetime, deadagi,
dial, dictate, digittimeout, directory, disa, dundilookup, eagi, endwhile,
execif, execiftime, externalivr, festival, getcpeid, gosubif, gotoif,
gotoiftime, hasnewvoicemail, hasvoicemail, iax2provision, ices, importvar,
lookupblacklist, lookupcidname, macroexit, macroif, mailboxexists,
meetmeadmin, milliwatt, mixmonitor, monitor, pickup, privacymanager,
readfile, realtime, realtimeupdate, responsetimeout, retrydial, ringing,
saydigits, sayphonetic, sayunixtime, sendimage, setcallerid,
setcdruserfield, setcidname, setcidnum, setrdnis, settransfercapabilit,
sipaddheader, sipdtmfmode, sipgetheader, stopmonitor, testclient,
txtcidname, vmauthenticate, voicemail, voicemailmain, wait, waitexten,
waitforring, waitforsilence, while)
Surely you can come up with a name slightly more descriptive -- maybe
idx?
Take pity on the next sod that has to plod through your dialplan. The
millisecond you spend typing a more meaningful name will be returned to
you (or your employer) a millionfold.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 7:36:54 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 09:22:25 Douglas Garstang wrote:
Yes, and by doing that your compounding the fact that all your variables
are global.
No, his variables are local to the channel he's using. Global variables are
a completely different beast.
--
Tilghman
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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] Asterisk as an IVR solution
On Friday 11 July 2008 12:07:37 Douglas Garstang wrote: A subroutine with arguments? In 1.6, yes, or in the 1.4 backport, yes. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote: I want to track call duration while the call is in progress. To accomplish what? Are you wanting to beep the channel every 10 seconds? Are you wanting to play a you have 60 seconds left message when they approach some quota? Are you wanting to limit the call to 5 minutes and 23 seconds? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune: Unable to set impedance
Tzafrir Cohen wrote So this is an FXS module. Guess I mixed it up ;-) For starters, do you have echo cancellation enabled? asterisk -rx 'zap show channel 120' | grep 'Echo' Echo Cancellation: 128 taps unless TDM bridged, currently OFF How can I turn it on? Udo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jul 2008 08:10:38 -0700
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound
but I cant hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
If Asterisk says it is playing the file, then I would suspect the file
itself has nothing to say. Try copying the file to your computer and
playing it. If it does indeed play locally on your computer with
audio, double check to make sure it is in the right format. I use AGI
to play files all the time. Actually, I use an AGI script as my whole
menu and dialing system to replace having to do it in AEL (so much
nicer to add a single MySQL record and suddenly have voicemail and
direct dial work instantly).
Daniel
I tested the files playing in other app, Winamp, and the file play fine.
I tested with other files ,sounds from asterisk, and I get the same thing.
In my spftphone doesnt hear anything
But this works
$AGI-say_number('9865')
so fine.
??
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera
de lo comun
*---*
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Re: [asterisk-users] Diagnosing dropped calls...
Unfounded rumors say that ABE doesn't come with app_rnddropcall ;-] On Fri, Jul 11, 2008 at 12:40 PM, Carlos Chavez [EMAIL PROTECTED] wrote: The other thing that baffles me about this setup is that it only seems to happen to people who are connected to the internal network in the office. They have about 30 remote users that have not reported this same problem, their issue is usually bandwidth related from their home connection. We have checked the internal network several times and there is not any obvious problem (apart from the dropped calls). They use high end Cisco switches and they were just audited to make sure there were no configuration errors. All the internal phones are Aastra (most are 9133i and some others 53i). On Thu, 2008-07-10 at 20:55 -0400, Steve Totaro wrote: Try dropping the IAX2 and only use SIP. Don't ask why? Just give it a try and see if things improve for you. Also when you assume, you make and ass out of you and me (just a little joke, get it? ass-u-me.) You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly (point to point) with G729 or just get a real ISDN or POTS lines. And then you will still have dropped calls depending on your volume and how vocal your users are. Usually, once they perceive a problem, then even if the other side of the call is on a cell and the cell drops the call, you will get a complaint. The only way to track those down are on a case by case basis with ANI II codes 61-63 http://www.nanpa.com/number_resource_info/ani_ii_assignments.html Thanks, Steve Totaro On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez [EMAIL PROTECTED] wrote: My customer has a 10mpbs fiber connection to the Internet so we have always assumed that the connection is not really a problem. We will look into it. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Friday 11 July 2008 12:40:47 Edwin Quijada wrote:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jul 2008 08:10:38 -0700
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound
but I cant hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
If Asterisk says it is playing the file, then I would suspect the file
itself has nothing to say. Try copying the file to your computer and
playing it. If it does indeed play locally on your computer with
audio, double check to make sure it is in the right format. I use AGI
to play files all the time. Actually, I use an AGI script as my whole
menu and dialing system to replace having to do it in AEL (so much
nicer to add a single MySQL record and suddenly have voicemail and
direct dial work instantly).
Daniel
I tested the files playing in other app, Winamp, and the file play fine.
I tested with other files ,sounds from asterisk, and I get the same thing.
In my spftphone doesnt hear anything
But this works
$AGI-say_number('9865')
so fine.
Check the format of the file. In most cases, the file should be 8000Hz,
single channel, uncompressed, signed linear, 16-bit samples format. Winamp
can play a great many different formats, but Asterisk is limited to the
formats for which it has a translator.
--
Tilghman
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Re: [asterisk-users] Tracking Call Time While in Dial()
Wanting to provide a real time call timer on a web page. - Original Message From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 10:17:01 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote: I want to track call duration while the call is in progress. To accomplish what? Are you wanting to beep the channel every 10 seconds? Are you wanting to play a you have 60 seconds left message when they approach some quota? Are you wanting to limit the call to 5 minutes and 23 seconds? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
From: Steve Edwards [EMAIL PROTECTED]
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Ugh. Yes, the variables are local to the current channel. However, they
are global to the entire dial plan within the current channel. I have
stepped on myself many times because I've had a loop counter called $i
for example, jumped somewhere else within that loop, reused the same
variable name, $i, and screwed up my logic.
Ugh indeed. While I sympathize with your local/global name space issues,
you lose credibility with your false economy.
IMNSHO, anybody who uses a single [common] letter for a variable
deserves a bump in the temperature when they reach their final resting
place :)
Surely you can come up with a name slightly more descriptive -- maybe
idx?
Take pity on the next sod that has to plod through your dialplan. The
millisecond you spend typing a more meaningful name will be returned
to you (or your employer) a millionfold.
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Fine, I'll call it ${LoopVariable} then... how's that going to fix the
problem?
It (obviously) doesn't. It just fixes the next guy's problem when he
tries to read your dialplan -- as stated above.
I'm just suggesting better practices. Even examples should demonstrate
best practices because they form the basis of some coders only source of
knowledge.
Kind of like not top posting in a list where the posted etiquette is
not top posting ;)
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] fxotune: Unable to set impedance
On Fri, Jul 11, 2008 at 07:24:23PM +0200, Udo Schacht-Wiegand wrote: Tzafrir Cohen wrote So this is an FXS module. Guess I mixed it up ;-) For starters, do you have echo cancellation enabled? asterisk -rx 'zap show channel 120' | grep 'Echo' Echo Cancellation: 128 taps unless TDM bridged, currently OFF It is enabled. 128 taps (16 ms) is the default value which is normally more than enough for FXS. It will bo on when a call will actually use an echo canceller. -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Fri, 11 Jul 2008, Tilghman Lesher wrote: On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote: My problem is that i cant hear anything when play the file sound using $AGI-stream_file($filename); I put asterisk in verbose mode but just see that it plays the sound but I cant hear anything. Check the format of the file. In most cases, the file should be 8000Hz, single channel, uncompressed, signed linear, 16-bit samples format. Winamp can play a great many different formats, but Asterisk is limited to the formats for which it has a translator. If the file is a wav, it should look something like this: -t2::sedwards:~$ file example.wav example.wav: RIFF (little-endian) data, WAVE audio, Microsoft\ PCM, 16 bit, mono 8000 Hz Also, just in case you trip over this, you pass a file name to Asterisk, not a file type -- the bit after the period. Asterisk chooses the best type from files of the same name based on the codecs available to the channel. 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri version 1.4.5 Released
The Asterisk development team has released version 1.4.5 of libpri. This release was made solely to correct a problem introduced in version 1.4.4. In February of 2008, a change was made in libpri to support inband audio (progress) when the far end of a PRI circuit issues a RELEASE message, indicating they want to terminate the call. This change was necessary for some applications where the telco providing the circuit wants to provide a 'release message' before actually hanging up the call. Unfortunately, many users have PRI circuits that are not compatible with this behavior, and this results in their PRI B-channels being left open for anywhere from 2 to 20 seconds (or more) before the calls are finally terminated. This version of libpri retains the ability to operate in this mode, but it is now a configurable option which defaults to being 'off'. The next releases of Asterisk will have configuration options to turn this behavior on if the user desires. Thanks for using libpri and Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd text in sip debug
I saw this shortly after ssh'ing into a box that was not answering sip inbound calls: --- SIP read from 192.168.100.253:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233 Max-Forwards: 70 From: xx sip: xx @192.168.100.5;tag=as588c6a60 To: sip:[EMAIL PROTECTED];tag=faLty Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 What does faLty mean? Ext 203 is a Snom M3 portable handset. Once I initiated the ssh session, I called and it answered my phone call, and the xx was my phone number calling in. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd text in sip debug
Joseph L. Casale wrote: I saw this shortly after ssh'ing into a box that was not answering sip inbound calls: --- SIP read from 192.168.100.253:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233 Max-Forwards: 70 From: xx sip: xx @192.168.100.5;tag=as588c6a60 To: sip:[EMAIL PROTECTED];tag=faLty Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 What does faLty mean? Ext 203 is a Snom M3 portable handset. Once I initiated the ssh session, I called and it answered my phone call, and the xx was my phone number calling in. Thanks! jlc The To: tag on the response is the tag generated by your phone. It is generated pretty much at random. It's just a happy coincidence that it happened to nearly spell the word faulty. Still, that's kind of funny though :) Nothing to worry about. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). The T-1's aren't on that, though, right? Next to that is a box with 4 slots for T1 cards, we used to have a T1 internet connection and its card is still in there. Slot 2 has the flex-grow T1 card in it. That's the smartjack, in the box. One of the pairs from the D-Marc goes into this T1 card and it Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. provides a RJ-45 connection for the T1 line that runs either to the Adtran or to our Digium T1 card. Probably an RJ-48, actually, but who's counting. :-) I hope that answers the question, as I am not entirely sure what a shelf or smartjack are. Though I will feel really stupid if you say a shelf is something you store stuff on. See above. :-) Cheers, -- jra -- 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. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No service on phones...
Today I had a problem where the internet connection is unstable so calls are getting dropped all over the place. The one thing I do not understand is that at least 30 phones on the internal network went to No Service. Since they are on the same network segment and on the same subnet I do not see why the Internet connection sould affect them. The asterisk server is behing NAT and we use externip=201.161.XXX.XXX for outside sip phones and providers. Any recommendations? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
Very cool, you've piqued my interest. Since I haven't launched an instance before, where's the best place to learn to do that? What's the approximate monthly cost of hosting an Asterisk PBX on EC2? Ronald Lewis wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTERISK/ENSWITCH ON EC2
Hi All, I seen earlier the first guide on deploying asterisk on EC2 in the list becoming available. Has anyone deployed a hosted environment like enswitch using EC2? I was wondering if anyone had any thoughts on concerns on the feasibility in doing this using cloud computing? I would have thought that allocating CPU resources and bandwidth dynamically would have been an issue for VoIP asterisk like media servers and gateways. From reading Amazons site, it appears that it is very easy to adjust the amount of CPU resources available to your virtual servers. For setting up a VoIP service provider and not having the headache of dealing with the hassle and expenses of hardware, racks, cages etc, it looks pretty attractive. Any thoughts? Robert McNaught ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd text in sip debug
Still, that's kind of funny though :) Hilarious :) This CentOS machine running asterisk is in a Xen vm and its not behaving well. I am moving it to physical hardware asap and thought that may have been part of some indication of the myriad of issues it has. That is a priceless coincidence! Thanks for the quick reply! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK/ENSWITCH ON EC2
Googlezon will rule the world. http://www.robinsloan.com/epic/ On Fri, Jul 11, 2008 at 3:28 PM, Robert McNaught [EMAIL PROTECTED] wrote: Hi All, I seen earlier the first guide on deploying asterisk on EC2 in the list becoming available. Has anyone deployed a hosted environment like enswitch using EC2? I was wondering if anyone had any thoughts on concerns on the feasibility in doing this using cloud computing? I would have thought that allocating CPU resources and bandwidth dynamically would have been an issue for VoIP asterisk like media servers and gateways. From reading Amazons site, it appears that it is very easy to adjust the amount of CPU resources available to your virtual servers. For setting up a VoIP service provider and not having the headache of dealing with the hassle and expenses of hardware, racks, cages etc, it looks pretty attractive. Any thoughts? Robert McNaught ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Jul 11, 2008, at 12:09 PM, Jay R. Ashworth wrote: On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). The T-1's aren't on that, though, right? ... Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. Yup. I thought that pretty funny myself. 10 year old analog wires running a digital T1. :) And they do have some caps on them, I think it was red but not 100% sure. I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. I called Verizon and opened a ticket for why those 2 lines are behaving differently and that sounds like the problem, but I won't know for sure until I can test and try calling on one of the lines that does toggle the RA bit. As soon as I get that tested I will report that, though I expect that should fix the hangup issue. Thanks, Daniel Cheers, -- jra -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote: Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. Yup. I thought that pretty funny myself. 10 year old analog wires running a digital T1. :) And they do have some caps on them, I think it was red but not 100% sure. No, that's not the unusual part. The unusual part is just that both analog and digital services are on the same block. Maybe it's a regional think... I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. I called Verizon and opened a ticket for why those 2 lines are behaving differently and that sounds like the problem, but I won't know for sure until I can test and try calling on one of the lines that does toggle the RA bit. As soon as I get that tested I will report that, though I expect that should fix the hangup issue. Aha! Good luck with that. Cheers, - jra -- 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. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
Date: Fri, 11 Jul 2008 11:29:58 -0700
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Fri, 11 Jul 2008, Tilghman Lesher wrote:
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename); I put asterisk in verbose mode
but just see that it plays the sound but I cant hear anything.
Check the format of the file. In most cases, the file should be 8000Hz,
single channel, uncompressed, signed linear, 16-bit samples format.
Winamp can play a great many different formats, but Asterisk is limited
to the formats for which it has a translator.
If the file is a wav, it should look something like this:
-t2::sedwards:~$ file example.wav
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft\
PCM, 16 bit, mono 8000 Hz
Also, just in case you trip over this, you pass a file name to Asterisk,
not a file type -- the bit after the period. Asterisk chooses the best
type from files of the same name based on the codecs available to the
channel.
vm-debian#file tts-hello
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
8000 Hz
I recorded the sound using Cepstral. This is my AGI
I thought maybe was my sound card but this works fine
$AGI-say_number('9865');
$AGI-say_digits('873746');
and I can hear it in my SIP phone
use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
#
$AGI-say_number('9865');
$AGI-say_digits('873746');
speak(Hello World);
sub speak
{
$text = $_[0];
my $hash = md5_hex($text);
my $ttsdir = /var/lib/asterisk/sounds/tts;
my $cepoptions = -p audio/sampling-rate=8000,audio/channels=1;
my $wavefile = $ttsdir/tts-$hash.wav;
unless (-f $wavefile)
{
open(fileOUT, /var/lib/asterisk/sounds/tts/say-text-$hash.txt);
print fileOUT $text;
close(fileOUT);
my $execf=/opt/swift/bin/swift -f $ttsdir/say-text-$hash.txt -o
$wavefile $cepoptions;
system($execf);
unlink($ttsdir/say-text-$hash.txt);
}
$filename = 'tts/'.basename('tts/'.basename($wavefile,.wav));
$AGI-stream_file($filename);
# unlink($wavefile);
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] No service on phones...
Carlos Chavez wrote: Today I had a problem where the internet connection is unstable so calls are getting dropped all over the place. The one thing I do not understand is that at least 30 phones on the internal network went to No Service. When this happens try to capture DNS traffic originating from your Asterisk. I suspect you will find Asterisk is going crazy trying to resolve DNS names and getting no response. I see that a lot. I usually brings down to a grind the whole pbx. Andres http://www.neuroredes.com Since they are on the same network segment and on the same subnet I do not see why the Internet connection sould affect them. The asterisk server is behing NAT and we use externip=201.161.XXX.XXX for outside sip phones and providers. Any recommendations? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
That happens all the time when the T1s are purchased from a CLEC as the RBOCs just deliver the clec pairs wherever. I can think of at least two or three demarcs that I have been to in the last few months that were mixed like that. Here in Qwest territory the T1s use a different color cross connect wire (red/blue and red/orange vs the yellow/blue that the analog lines use). John van Oppen Spectrum Networks LLC 206.973.8302 (Direct) 206.973.8300 (main office) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Friday, July 11, 2008 1:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] US T1 Hangup Detection On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote: Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. Yup. I thought that pretty funny myself. 10 year old analog wires running a digital T1. :) And they do have some caps on them, I think it was red but not 100% sure. No, that's not the unusual part. The unusual part is just that both analog and digital services are on the same block. Maybe it's a regional think... I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. I called Verizon and opened a ticket for why those 2 lines are behaving differently and that sounds like the problem, but I won't know for sure until I can test and try calling on one of the lines that does toggle the RA bit. As soon as I get that tested I will report that, though I expect that should fix the hangup issue. Aha! Good luck with that. Cheers, - jra -- 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. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on the T1 wires.
Yup. I thought that pretty funny myself. 10 year old analog wires
running a digital T1. :) And they do have some caps on them, I think
it was red but not 100% sure.
No, that's not the unusual part. The unusual part is just that both
analog and digital services are on the same block. Maybe it's a
regional think...
That's really not unusual. It's not /preferred/, but that's an entirely
different can of worms.
In general, if copper is available into a building, the telco is going to
look very seriously at the possibility of using that. If the building is
already wired and the copper tests clean, the telco will want to use that.
In most existing situations, that will already be terminated in a can with
lightning suppression and will have been crossed over to RJ21X's that are
going to whatever suites are in the building.
Since the telco will have /no/ /problem/ running the T1 over their outside
plant and up to the can on what is approximately Category 3 wire, and the
T1 signal is going to have been running alongside those same analog wires
for probably a few miles, what happens next should be obvious.
Suite 214 wants a T1. There's already a 25-pair going up there from the
RJ21X. It's second story, so do you go and spend an {hour, afternoon,
etc} figuring out how to run fresh wire, or do you notice that only 6 pair
are in use on the RJ21X, and decide to feed up on the existing cable?
Now, if you're nasty and you don't separate it (typically I see the bottom
used for data) and you don't put redcaps on, yeah, then that is just
looking for eventual trouble. And who knows, the wire may be cruddy, so
maybe you still end up doing the separate run. But it probably works.
I've seen this often enough. Would I prefer to see new cable run? Sure.
But we've all done our copper sins. I've seen a lot of things that are
uglier than that. Here's one of them:
http://www.sol.net/hallofshame/
(I've always meant to expand that page, but it seems that I never get the
good photos of bad stuff)
Lack of space, lack of need, lack of having another RJ21X in the truck are
just a few other obvious reasons that this might be done.
... JG
--
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 11, 2008, at 1:31 PM, Edwin Quijada wrote: vm-debian#file tts-hello example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Other than the filename being wrong which I would assume is the result of a copy and paste from the original e-mail, that looks right. Can you paste the asterisk log section around where it is playing the file, including the line that shows it playing? Something in the log may give a clue. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MagicJack quality
I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Fri, 11 Jul 2008 17:13:15 -0400, C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? More memory CPU power then the average hard phone? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Good light codecs like speex, and minimal feature sets. C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Not sure about magicjack but skype has supernodes that play a large part in how the system works well. http://geemodo.blogspot.com/2006/10/dont-be-skype-supernode-or-how-not-to.html Thanks, Steve T On Fri, Jul 11, 2008 at 5:29 PM, Anthony Francis [EMAIL PROTECTED] wrote: Good light codecs like speex, and minimal feature sets. C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Better handling of the packets, that's for sure. Also, the algorithm is smart, and flexible... that being said, it opens more questions than answers. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, July 11, 2008 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality On Fri, 11 Jul 2008 17:13:15 -0400, C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? More memory CPU power then the average hard phone? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Thanks, Steve T On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich [EMAIL PROTECTED] wrote: Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Fri, 11 Jul 2008 18:28:09 -0400, Steve Totaro wrote: I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Actually I get a steady stream of MJ traffic to my blog for a most unexpected reason. There is an unofficial MJ forum. On that forum there is a bunch of information about adapting the HP T5700 series thin clients to achieve a dedicated host for MJ. That way people can keep their MJ operational without leaving their PC on all the time. I wrote a series of posts about using T5700s for a variety of tasks, and the MJ forum points to those posts. The people on the forums have come up with hacks to the XPe OS to insert the MJ drivers and other utilities. There's even one guy selling mod'ed thin clients for those who lack the DIY desire or skills. Her'sa link: http://unofficialmagicjack.forum2u.org/hp-thin-client-modifications-upgr ades-for-magicjack-t1493.html It's a pretty cool example of a community springing up around a service. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
As per the ads, if people ignore them or not, doesn't matter. Advertisers will fall in love with the idea that the venue reaches 1 million people, or more. As per the price of the service, they might be calculating the fact that the average monthly consumption of minutes on a softphone could be lower that the average monthly consumption on hardphones. After all, having to have that cpu on to make the call, is a drag. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, July 11, 2008 6:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Thanks, Steve T On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich [EMAIL PROTECTED] wrote: Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recharge Dial Limit....?
Here's an interesting challange. I need to implement a calling card application, where I call the Dial() command and pass it (L)imit information. Nothing difficult about that. Except it is a requirement that rather than ending the call when the limit is reached, the user gets the option to recharge their account. Now, since the dial() command will just end the call when the limit has been reached, how could I possibly do this? The only way I can think of is to have another system send Asterisk a SIP reinvite before the call ends, and direct the media somewhere else so that we can drop into a new IVR and let them top off their account. A reinvite would have to go to the remote party too, so that they could listen to music on hold while the caller was topping off their account. It just occurred to me that this may not work. The (L)imit information passed to the Dial application has not changed. The Dial() application would still end the call. Ideas? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
As Michael Graves points out, people will hack it to run on thin clients and why not virtual machines with very limited access? Maybe an AP with a USB port and OpenWRT or something? Remember when NetZero really cost nothing? I had a program someone wrote to close the as Windows, later I figured out that if I removed a dll file and then just put some junk in a text file and name it whateverthefilewas.dll NetZero was free and I didn't have to use any programs to close the ad windows. Remember the free webhosts that put ads at the bottom of your page but you did get decent free hosting. Remember the scripts that came out within weeks that eliminated those ads? Thanks, Steve T On Fri, Jul 11, 2008 at 6:53 PM, C. Savinovich [EMAIL PROTECTED] wrote: As per the ads, if people ignore them or not, doesn't matter. Advertisers will fall in love with the idea that the venue reaches 1 million people, or more. As per the price of the service, they might be calculating the fact that the average monthly consumption of minutes on a softphone could be lower that the average monthly consumption on hardphones. After all, having to have that cpu on to make the call, is a drag. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, July 11, 2008 6:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality I don't see Magicjack being around long. The business model isn't sustainable without tons of ads, and even then, people will either ignore them if they are audio or if they are popups, they will simply close them or disable them. I might buy one just to hack it. Has anyone sniffed it or poked around at all on lists? Thanks, Steve T On Fri, Jul 11, 2008 at 6:03 PM, C. Savinovich [EMAIL PROTECTED] wrote: Yes, I have designed two different webphones, granted, using third party libraries, and magicjack's quality is better. I acknowledge that. Thank you, but referring me to someone's review won't help me much... I am interested in the internals. Regardless, their technique has a twist, and I am a naturally very curious *technical* fellow. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Friday, July 11, 2008 5:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MagicJack quality I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Recharge Dial Limit....?
On Fri, Jul 11, 2008 at 7:12 PM, Douglas Garstang [EMAIL PROTECTED] wrote: Here's an interesting challange. I need to implement a calling card application, where I call the Dial() command and pass it (L)imit information. Nothing difficult about that. Except it is a requirement that rather than ending the call when the limit is reached, the user gets the option to recharge their account. Now, since the dial() command will just end the call when the limit has been reached, how could I possibly do this? The only way I can think of is to have another system send Asterisk a SIP reinvite before the call ends, and direct the media somewhere else so that we can drop into a new IVR and let them top off their account. A reinvite would have to go to the remote party too, so that they could listen to music on hold while the caller was topping off their account. It just occurred to me that this may not work. The (L)imit information passed to the Dial application has not changed. The Dial() application would still end the call. Ideas? Doug. Use an AGI, dissect ASTCC or ASTPP AGIs, all the goodies you want are in there. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recharge Dial Limit....?
Thanks, but how does that extend the core functionality of Dial()? If Dial() can't do it, how does a wrapper do it? - Original Message From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 4:29:50 PM Subject: Re: [asterisk-users] Recharge Dial Limit? On Fri, Jul 11, 2008 at 7:12 PM, Douglas Garstang [EMAIL PROTECTED] wrote: Here's an interesting challange. I need to implement a calling card application, where I call the Dial() command and pass it (L)imit information. Nothing difficult about that. Except it is a requirement that rather than ending the call when the limit is reached, the user gets the option to recharge their account. Now, since the dial() command will just end the call when the limit has been reached, how could I possibly do this? The only way I can think of is to have another system send Asterisk a SIP reinvite before the call ends, and direct the media somewhere else so that we can drop into a new IVR and let them top off their account. A reinvite would have to go to the remote party too, so that they could listen to music on hold while the caller was topping off their account. It just occurred to me that this may not work. The (L)imit information passed to the Dial application has not changed. The Dial() application would still end the call. Ideas? Doug. Use an AGI, dissect ASTCC or ASTPP AGIs, all the goodies you want are in there. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. There's already 4 public images on ec2 mentioning Asterisk in their names so wouldn't it be easier to try out one of those rather than install all the bits and pieces on a base Linux image? An interesting paper on ec2 and Asterisk would be one that discusses what the call quality is like from both inside and outside the US. When I briefy ran up an instance at this time last year it actually seemed ok. From a provider's point of view running Asterisk on the ec2 cloud does pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be 3.6GB/hour or 86.4GB/day. At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The server instance will cost you $72/month so total cost for 100/calls per month is $332. A typical dedicated server for $300/month is roughly equivalent to an ec2 small instance and comes with 500GB of bandwidth/month which is only a fifth of what's required but you could probably get the extra 2TB/month thrown in for $32/month making the dedicated server and ec2 prices the same. There are serious pros and cons between these approaches. With the ec2 you don't get a permanent static IP, with a dedicated server you do. With ec2 you could scale up and down between 1 server and 4 servers at the drop of a hat to save costs and cope with peak and quite times, with dedicated servers you're stuff with 12 or 24 month contracts for the number of servers you'd need under maximum load. And then of course the major factor for both is what the call quality will be like. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited. CPU Intel Xeon X5355 1x 4x 2.66 GHz L2: 8Mo, FSB: 1333MHzQuadruple Coeur Architecture64 bits RAM 8 Go FBDIMM DDR2 HDD 2x 750 Go Type HDD SATA2 RAID HARD 1Interfaces 2 x 1 Gbps SPEED 2 Gbps Traffic UNLIMITED IP fixe2 adresses IP Fail-over http://www.ovh.com/fr/items/ip_failover.xml+8 adressesVPS Ready http://www.ovh.com/fr/produits/offres_vps.xml[image: Oui]IP Fail-over VPS64 IP (/26)IP enregistrées RIPE[image: Oui] Sauvegarde FTP http://www.ovh.com/fr/items/sauvegarde_ftp.xml750 Go 2008/7/12 Grey Man [EMAIL PROTECTED]: On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. There's already 4 public images on ec2 mentioning Asterisk in their names so wouldn't it be easier to try out one of those rather than install all the bits and pieces on a base Linux image? An interesting paper on ec2 and Asterisk would be one that discusses what the call quality is like from both inside and outside the US. When I briefy ran up an instance at this time last year it actually seemed ok. From a provider's point of view running Asterisk on the ec2 cloud does pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be 3.6GB/hour or 86.4GB/day. At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The server instance will cost you $72/month so total cost for 100/calls per month is $332. A typical dedicated server for $300/month is roughly equivalent to an ec2 small instance and comes with 500GB of bandwidth/month which is only a fifth of what's required but you could probably get the extra 2TB/month thrown in for $32/month making the dedicated server and ec2 prices the same. There are serious pros and cons between these approaches. With the ec2 you don't get a permanent static IP, with a dedicated server you do. With ec2 you could scale up and down between 1 server and 4 servers at the drop of a hat to save costs and cope with peak and quite times, with dedicated servers you're stuff with 12 or 24 month contracts for the number of servers you'd need under maximum load. And then of course the major factor for both is what the call quality will be like. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing dropped calls...
Quote Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly If your business lives and dies, get that regular, boring, RELIABLE, TDM-T1. SIP/VOIP/Whatever - Cool fun, great when it works TDM-T1 - Unsurpassed reliabilty Steve Totaro wrote: Unfounded rumors say that ABE doesn't come with app_rnddropcall ;-] On Fri, Jul 11, 2008 at 12:40 PM, Carlos Chavez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The other thing that baffles me about this setup is that it only seems to happen to people who are connected to the internal network in the office. They have about 30 remote users that have not reported this same problem, their issue is usually bandwidth related from their home connection. We have checked the internal network several times and there is not any obvious problem (apart from the dropped calls). They use high end Cisco switches and they were just audited to make sure there were no configuration errors. All the internal phones are Aastra (most are 9133i and some others 53i). On Thu, 2008-07-10 at 20:55 -0400, Steve Totaro wrote: Try dropping the IAX2 and only use SIP. Don't ask why? Just give it a try and see if things improve for you. Also when you assume, you make and ass out of you and me (just a little joke, get it? ass-u-me.) You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. Seriously though, if your business lives and dies by the phone system, get T1 with SIP from your provider directly (point to point) with G729 or just get a real ISDN or POTS lines. And then you will still have dropped calls depending on your volume and how vocal your users are. Usually, once they perceive a problem, then even if the other side of the call is on a cell and the cell drops the call, you will get a complaint. The only way to track those down are on a case by case basis with ANI II codes 61-63 http://www.nanpa.com/number_resource_info/ani_ii_assignments.html Thanks, Steve Totaro On Thu, Jul 10, 2008 at 7:15 PM, Carlos Chavez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My customer has a 10mpbs fiber connection to the Internet so we have always assumed that the connection is not really a problem. We will look into it. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ?Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Could you clarify how you end up with 1.4 Backport ? If you go to DIGIUM and download 1.4 do you have a backport 1.4 or is there a super-secret-non-more-secret-archive one would get it from ? I have never really understood this. Thank You Tilghman Lesher wrote: On Friday 11 July 2008 12:07:37 Douglas Garstang wrote: A subroutine with arguments? In 1.6, yes, or in the 1.4 backport, yes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
On Sat, Jul 12, 2008 at 2:16 AM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited. And I'm sure there will be someone somewhere who has a better deal as well. Whenever hosting gets mentioned in tech forums you always end up with lots of opinions (or pitches depending on how well they are disguised). My post was more in the vein of ballpark figures to compare the technincal merits of a grid approach, in this case manifested by Amazon's ec2, to a dedicated server approach for running a VoIP provider service using Asterisk. It's probably already gone off the OP's original topic enough now that they'll be people jumping up and down that this should be on the Biz list... Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS I don't know what Magic-jack does (I've never actually seen one), but I know the key thing about Skype that impresses people - its wideband voice codec. A lot of people poo-poo the idea that wideband voice has value in a phone call. They are either close to deaf, or have never tried it. Clarity is profoundly improved. Skype seems to use various tricks to keep the packet flow smooth, but its wideband that makes it sound better than the PSTN. You might think a standard phone plugged into an adaptor, like a Magic-jack, would be limited to narrow band voice, as that is all the phone was designed for. It turns out most phones only aggressively filter at the low end of the band. They let a lot of energy above 4kHz through, and they do generally sound better through a wideband codec. Many modern line interface chips are actually capable of running in a 16k samples/second mode, even though most are programmed for 8k samples/second. I think the ones on the TDM400P type cards can. Some from Silicon Labs certainly can, and chips from Zarlink and others can. If Magic-jack sounds impressively clear, a wideband codec would be my guess. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Douglas Garstang wrote: Well, a macro is the closest thing the dial plan has to a subroutine, and without that, we might as well be programming in Assembler (no subroutines, local variables, lots of goto's... sound familiar?). Doug. - Original Message From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 11, 2008 7:20:40 AM Subject: Re: [asterisk-users] Asterisk as an IVR solution On Friday 11 July 2008 01:28:34 Douglas Garstang wrote: Well I can tell you that it makes a difficult programming environment, just a little more difficult. It means I can't implement a menu as a single reusable piece of code inside a macro. That's the point. A Macro is NOT a subroutine. It's like saying that you can't effectively hammer a nail with a screwdriver, and therefore you think the screwdriver has a known problem. There's nothing wrong with the screwdriver; it simply is the wrong tool for the job. I must somewhat disagree with you on this. 1) A MACRO could reasonably viewed as the Current Context, so if the jumping/branching from extension to extension that takes place in other contexts, it would if fact be quite reasonable and expected that this would happen in a MACRO. 2) As SUBROUTINES did not come standard until 1.6, it might be reasonably stated that no appropriate tool existed until 1.6, and since good programming practice uses subroutines, and a MACRO did not work like subroutine, even though it LOOKS like one, people are not fully happy that the closest tool they had, did not do the job Just a thought , no flame intended or implied. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
Thank You - clears up a LOT I did not fully grasp Tilghman Lesher wrote: On Friday 11 July 2008 01:05:22 Al Baker wrote: Tilghman Lesher wrote: On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote: It's a known problem. If you call Background() in a macro, then Asterisk will look for the extensions to jump to in the CALLING Macro/context and NOT the Macro that the Background() app was called in. I wouldn't call it a known problem. It works precisely as it was designed to work. It may not work the way that you want it to, but it works like a Macro: an independent set of instructions, with substitution, that acts as if it were invoked inline with the calling location. That is why Background will match in the context of the calling location: it acts like it never left that original context (and, in a way, it really didn't). Subroutines are a different beast, and they are available with the Gosub/ Return set of routines in app_stack.so. SO does that mean that if he used BACKGROUND is a SubRoutine he would get the correct or desired action , from his point of view? It would jump to the 1 Extension in the SUBROUTINE ? Yes, if he used Background within a Gosub, it would behave the way that he expects. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Fri, 11 Jul 2008, Edwin Quijada wrote: I recorded the sound using Cepstral. This is my AGI I thought maybe was my sound card but this works fine Why would you think it was the sound card? 1) Try enabling AGI debugging. For 1.2, enter agi debug and then execute your agi. The important part should look like: AGI Rx STREAM FILE /var/lib/asterisk/tts/tts-5f45f4c8732d220207032bb30a4398ee AGI Tx 200 result=0 endpos=9072 Note there is no file type and the (the escape digits) at the end of the Rx line. If you get something different, please post it. 2) Are your file ownership and permissions OK? Try: ls -dl /var/lib/asterisk/sounds/tts/ ls -l /var/lib/asterisk/sounds/tts/ 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 21:24:10 Al Baker wrote: Tilghman Lesher wrote: On Friday 11 July 2008 12:07:37 Douglas Garstang wrote: A subroutine with arguments? In 1.6, yes, or in the 1.4 backport, yes. Could you clarify how you end up with 1.4 Backport ? If you go to DIGIUM and download 1.4 do you have a backport 1.4 or is there a super-secret-non-more-secret-archive one would get it from ? I have never really understood this. The backport exists on svncommunity: http://svncommunity.digium.com/view/app_stack/1.4/ I also have two other repositories on svncommunity, both for backports: http://svncommunity.digium.com/view/func_odbc/1.4/ http://svncommunity.digium.com/view/tilghman/branches/1.4/ -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP Storage Problem
Hi,
I'm having a problem with IMAP storage and asterisk. Here is the error
message I get (in this instance its checking messages):
[Jul 11 23:14:12] WARNING[9888]: app_voicemail.c:8738 mm_log: IMAP
Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP protocol error: Authentication aborted
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:8741 mm_log: IMAP
Error: IMAP Authentication cancelled
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:4790 init_mailstream:
Can't connect to imap server
{mail.host.com:143/imap/notls/user=bigtizzies}INBOX
[Jul 11 23:14:12] ERROR[9888]: app_voicemail.c:2486 messagecount: IMAP
mailstream is NULL
voicemail.conf:
[general]
imapserver=mail.host.com
imapport=143
imapflags=notls
[default]
20002 = 1234,Sue's Mailbox,,,imapuser=bigtizzies|imapsecret=largedillas
Yet, when doing a 'mtest' (from the uw-imap directory I used for
asterisk) with {mail.host.com:143/imap/notls/user=bigtizzies}INBOX
and it works fine.
I seen a post on the Digium forums
(http://forums.digium.com/viewtopic.php?t=14432highlight=imap) where
another person had this same problem and he said he fixed it by fixing
a typo -- I've looked over my config and all seems good.
I'm attempting to connect to dovecot, here is a snip of the log on the
IMAP server:
Jul 11 23:26:04 esdiaz dovecot: imap-login: Aborted login (1
authentication attempts): method=PLAIN, rip=10.100.100.100,
lip=207.73.29.38
Anyone else ran across something like this? Ideas?
Thanks,
Marc
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Re: [asterisk-users] MagicJack quality
Steve Underwood wrote: C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS I don't know what Magic-jack does (I've never actually seen one), but I know the key thing about Skype that impresses people - its wideband voice codec. A lot of people poo-poo the idea that wideband voice has value in a phone call. They are either close to deaf, or have never tried it. Clarity is profoundly improved. Skype seems to use various tricks to keep the packet flow smooth, but its wideband that makes it sound better than the PSTN. You might think a standard phone plugged into an adaptor, like a Magic-jack, would be limited to narrow band voice, as that is all the phone was designed for. It turns out most phones only aggressively filter at the low end of the band. They let a lot of energy above 4kHz through, and they do generally sound better through a wideband codec. Many modern line interface chips are actually capable of running in a 16k samples/second mode, even though most are programmed for 8k samples/second. I think the ones on the TDM400P type cards can. Some from Silicon Labs certainly can, and chips from Zarlink and others can. If Magic-jack sounds impressively clear, a wideband codec would be my guess. Regards, Steve Like I said, Speex. It features Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
