RE: [Asterisk-Users] Snom 320 and message retrieve key
I expect the issue is the same problem we have with the 360's.
Quick fix is add the old Snom MWI fix to your dial plan but its not perfect
solution for us
as all our phones with DDI present 6 digits and we have already created our
mailboxes to match
the 3 digit ext number which means the users have to enter their mailbox number
as well as password.
exten = asterisk,1,Voicemail_blah(${CALLERIDNUM})
As you linked this appears to be an * bug.
Hope this helps
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek
Sent: 22 January 2006 21:19
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Snom 320 and message retrieve key
Hi,
I found some issues with Snom 320 message retrieve key. This button works only
when the MWI does not blink! If MWI
blinks and I do press retrieve button I get Unknown on display and busy
tone. From the sip debug it looks like that Snom
does not send credentials to Asterisk which responds with 407 Proxy Auth
required.
I have loaded Snom with latest 5 firmware. No change.
I'm using Asterisk 1.0.9 and have not tried 1.2.X.
Looks like this issue is related to http://bugs.digium.com/view.php?id=4801?
Does someone get Snom 320 retrieve button working with Asterisk 1.0.9?
Thanks,
-
David Hajek
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Re: [Asterisk-Users] Re: Polycom FW
On 1/23/06, Doug Lytle [EMAIL PROTECTED] wrote: Further, Polycom SIP phones have the longest boot time of any phone I've ever seen (something like 5 min, compared to a Sipure, less than Give a SIP based Cisco 79XX phone a try, just about as long in boot time. Huh? My 7905 takes well under 10 seconds, including Asterisk registration and NTP update. Granted, if it were DHCP it might take marginally longer, but 5 _minutes_? Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?
I recommend to use the mass deployment feature to maintain your phones. http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5 Besides this setting you cannot expect each setting is set by default to exactly match your needs. Different environments different setup. Best regards, Sven On Saturday 21 January 2006 19:57, Colin Anderson wrote: I can confirm that this is the issue. I now have to toggle it off manually on 120 phones. I can tell you, in the real world, you don't hand out passwords to users for their phones, they will not understand why you need a password for a phone. You may want to consider changing the default settings. Thanks for the info. Quick and accurate responses like yours are why I am a Snom fan. -Original Message- From: Christian Stredicke [mailto:[EMAIL PROTECTED] Sent: Saturday, January 21, 2006 8:51 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? The idea was that passwords will not be provisoned automatically, you must enter them manually on the phone. Which makes sense in scenarios where you completely automatically provision phones and hand out the password to the users. But maybe you are right, we should turn this off by default. I also had some pain with it! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: Saturday, January 21, 2006 10:46 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? Christian, Why is this this setting on by default? I don't understand why anyone would want this behavior. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Christian Stredicke [mailto:[EMAIL PROTECTED] Sent: Friday, January 20, 2006 8:05 PM To: Colin Anderson Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? Did you try to turn Challenge Response on Phone off in the advanced settings on the web interface? CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, January 20, 2006 8:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] OT:Snom 360 prompt for registration pwd? I have a whack of Snom 360's. Occasionally, *some* of them, prompt the user, on the screen, for the registration password. You enter it, everything's OK. Next day, same thing. This is like on 5 or 6 phones out of a lot of 120. It's *always* the same phones. I haven't drilled down to things like firmware rev yet, since I ordered them all as one lot, but I'm wondering if anyone knows under which circumstances a 360 would forget it's reg password? tia ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- See our FAQs at: http://www.snom.com/faq0.html?L=1 Whitepapers at: http://www.snom.com/white_papers.html --- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.com --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How to disable WARNINGS in CLI
thanks buddyOn 1/23/06, Cameron Grant [EMAIL PROTECTED] wrote: check /etc/asterisk/logger.confregards,cameron -- Forwarded message -- From: Angelito Manansala [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 22 Jan 2006 06:57:05 +0800 Subject: [Asterisk-Users] How to disable WARNINGS in CLI Hi guys, anyone knows how to disable the WARNINGS in cli, i set verbose 0 but the warning still show.. Thanks, Lito___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128 msn: [EMAIL PROTECTED]skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] When/whether to use SER?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a small simple rule to start with! Di you whant to have peiple calling in with sip to you domin then you have to use SER as a SIP server, and ser as a connection to the telco world. If you are only using the phones for PBX uses - use only Asterisk My 5 cents. best regards jan Steven wrote: I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I have yet to open my server to the Internet to be accessible to travelers without VPN first. I have done some testing with VOIP provider though my firewall to FWD and VOIPSTUNT. Where might SER help? Why are people using it with Asterisk? - -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFD1JZ4QEpdoflEoIsRAocNAKDJwx0pyB3Y1w2hVqRFxIh1An77jQCg61IZ emsPYxXDxM1gYeeCM8L/6VU= =fG5l -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-1.2.1.tar on Suse Linux 9 (Atif Nadeem)
Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
Hi All, I would like some clarification about licensing. Does this non-commericial license provide me for usage inside my company (We're not telephony provider so we are not using telephony services for making money). As I understood from license agreement, I cannot use it? Regards and thank you all, Dusko -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, January 23, 2006 12:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2? I also didnt comment on whether or not anyone can prove that you do have licenses, even if they know you use the codecs. Because to rely on that would be dubious at best, shut you down at worst. Out of curiosity, I wonder what one's legal position would be if one bought the appropriate number of licences from Digium, yet used the gcc (or intel) compiled binaries. Why would you want to do that, you ask? Well, for the same reasons why people buy games then apply no-CD cracks to them anyway - convenience. If one wanted to shift g729 licences around from machine to machine on a frequent basis, the mac-based licencing method might prove cumbersome. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing the none commercialintelg729codecsinto [EMAIL PROTECTED] 2.2?
On Mon, 2006-01-23 at 09:57 +0100, Dusko Tubin wrote: Hi All, I would like some clarification about licensing. Does this non-commericial license provide me for usage inside my company (We're not telephony provider so we are not using telephony services for making money). As I understood from license agreement, I cannot use it? Regards and thank you all, Dusko You are now asking for very specific legal advice for a very specific situation. I will speak about generalities but wont about specifics - the reason is that if its a general legal concept and not specific advice I cant be held liable (usually anyway). I would however advise you to seek professional legal counsel for a specific legal issue such as this. You also havent given anywhere near enough information for anyone to answer the question with anything other than 'maybe' - note this is not a solicitation for more info, just a reason that no one can give you anything that could be accurate unless they think the laws where they are apply globally :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set-up LCR
How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up maintain LCR? c. multiple connection to one gateway? Example: +886223456789could be reachable via a. ENUM free b. Dundifree c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking for a way, that covers all above. Voipstunt would get multiple accounts, whereby some are ONLY for free calls, and the others for paid connection. That way I do not need to check all the time the free connection. Has anybody done already some parts of that? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xlite set-up program
I am looking for a way to signup users and provide them with a file which includes all settings, just to put somewhere. Does something like that exist? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dnid support?
*bump* Anyone? I still can't find little/no info on DNID... :-/ Regards, Evert Evert Meulie wrote: Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 1 - ext. 1 913 - 2 - ext. 2 913-1 913-2 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the dialparties.agi script. This script sees and identifies the correct dnid, but I am having some trouble to get the dialplan to act on this value. The info in the Wiki ( http://www.voip-info.org/tiki-index.php?page=DNID ) is not of much help either. Anyone here with any suggestions? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi Examples
On Fri, Jan 20, 2006 at 09:20:43PM -0500, Michael Miller wrote: I have over 50 Asterisk servers geographically distributed in pairs all connected via DUNDi. Contact me off list and I will be happy to describe my experience. I'm also interested in knowing more of this. Why not write to the list so that more people may know about it? Regards, Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set-up LCR
Ronald Wiplinger a écrit : How to set-up LCR ? Easy! sudo perl -MCPAN -e 'install Asterisk::LCR' Then create a directory in which to work in, such as: mkdir /tmp/lcr Once you're in this directory, create a config file such as: [comparer] package = Asterisk::LCR::Comparer::XERAND currency = eur [dialer] package = Asterisk::LCR::Dialer::MinCost locale = fr [import:voipjet] package = Asterisk::LCR::Importer::VoIPJet dial = us IAX2/[EMAIL PROTECTED]/REPLACEME [import:nufone] package = Asterisk::LCR::Importer::NuFone dial = us IAX2/[EMAIL PROTECTED]/REPLACEME Then run successively: asterisk-lcr-import myconfig.cfg asterisk-lcr-build myconfig.cfg asterisk-lcr-dialplan myconfig.cfg lcr.conf move lcr.conf in /etc/asterisk, and include it in your existing dial plan using: #include lcr.conf a. which companies can be used with LCR? At the moment, NuFone, PlainVoIP, RichMedium and VoIPJet are supported. Virtually any company which offers publically downloadable CSV rate files can be added. If you know anymore downloadable CSV files please tell me and I will add it to the program. b. how to set-up maintain LCR? Repeat steps above, reload dialplan. c. multiple connection to one gateway? Example: +886223456789could be reachable via a. ENUM free b. Dundifree c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ Write a custom CSV file for your free connections and an import module for Asterisk::LCR. You can find the docs here: http://search.cpan.org/~jhiver/Asterisk-LCR-0.06/lib/Asterisk/LCR.pm Feel free to peek at the code. Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Odd you should have this problem as I had exactly the same. In my case it was a slow DHCP server. Around 7 seconds in the phones tries to time sync. If the phone hasn't got an IP address then this time sync fails but it doesn't retry. I emailed Grandstream about it but got nowhere. I changed my DHCP server from Windows to Linux and now DHCP is much faster and time sync is working. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: 21 January 2006 06:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP On Dec 31, 2005, at 7:28 AM, Ross C wrote: Peter, After upgrading to 1.0.1.13 I had some miscellaneous problems on one of my GXP-2000's--it would grab an IP address, but it wouldn't get the time/date, it wouldn't register, blah blah blah. I could access the web interface OK, so it wasn't a network issue (I don't think). Anyway...I ended up resetting to factory defaults and all is well now. Maybe try that? That has solved some other problems I've had as well. I just got a 2000 which does exactly this (our first for evaluation.. which is somewhat disappointing thus far). I could see in a packet sniffer a weird cycle of DHCP requests like it got an IP but kept retrying? A power cycle doesn't solve the problem (it's had many, and dozens of software resets). A reset with the MAC input doesn't work either for me. The phone was at an older FW when I got it (ending in .9, I think) and then updated to to the latest stable (.12 I think off the top of my head). Btw- the firmware update was a pain. HTTP updates were hitting the server (Apache) with 'bad request' results. I needed to set up my own tfpt server to make it work. Off lan updates weren't working, either, in any case. The phone will register and work when it has a static address assigned, but not when set for DHCP. In all cases, the clock is always wrong. I can see with a packet sniffer that the NTP request is sent and received, but with no effect on the phone display. Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Thanks! TGIF! :') Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling zaptel
It's been about 2 months since I have updated my asterisk box. I was running CVS HEAD and I notice a whole lot has changed since my last update! I'm running Debian Sarge up to date on a 2.4 Kernel. I was updating about every 2 or 3 weeks and never had any problems compiling zaptel/libpri/asterisk I now am coming out of a deep sleep and want to get back into it again, but zaptel will not compile. Is this a bug that I just need to wait for to be fixed? Or am I totally missing something that I need to make it compile? Followed the intructions on the site... Also noticed that everything now uses svn instead of CVS. Here's what I get when I try to compile zaptel: make install cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -lm gendigits.c -o gendigits ./gendigits gcc -I/lib/modules/2.4.27-2-386/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.27-2-386/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.27-2-386/build/drivers/net/wan -I/lib/modules/2.4.27-2-386/build/include/net -DMODVERSIONS -include /lib/modules/2.4.27-2-386/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c zaptel.c:41:21: version.h: No such file or directory In file included from /lib/modules/2.4.27-2-386/build/include/linux/spinlock.h:6, from /lib/modules/2.4.27-2-386/build/include/linux/module.h:12, from zaptel.c:45: /lib/modules/2.4.27-2-386/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.27-2-386/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.27-2-386/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zaptel.c: In function `zt_ctl_ioctl': zaptel.c:3476: error: `ZAPTEL_VERSION' undeclared (first use in this function) zaptel.c:3476: error: (Each undeclared identifier is reported only once zaptel.c:3476: error: for each function it appears in.) zaptel.c: In function `zt_init': zaptel.c:6838: error: `ZAPTEL_VERSION' undeclared (first use in this function) make: *** [zaptel.o] Error 1 Any ideas what I should try next? Thanks! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error compiling zaptel
Bummer - Possibly a bug The stable stuff compiles and runs fine :( Steve - It's been about 2 months since I have updated my asterisk box. I was running CVS HEAD and I notice a whole lot has changed since my last update! I'm running Debian Sarge up to date on a 2.4 Kernel. I was updating about every 2 or 3 weeks and never had any problems compiling zaptel/libpri/asterisk I now am coming out of a deep sleep and want to get back into it again, but zaptel will not compile. Is this a bug that I just need to wait for to be fixed? Or am I totally missing something that I need to make it compile? Followed the intructions on the site... Also noticed that everything now uses svn instead of CVS. Here's what I get when I try to compile zaptel: make install cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -lm gendigits.c -o gendigits ./gendigits gcc -I/lib/modules/2.4.27-2-386/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.27-2-386/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.27-2-386/build/drivers/net/wan -I/lib/modules/2.4.27-2-386/build/include/net -DMODVERSIONS -include /lib/modules/2.4.27-2-386/build/include/linux/modversions.h -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c zaptel.c:41:21: version.h: No such file or directory In file included from /lib/modules/2.4.27-2-386/build/include/linux/spinlock.h:6, from /lib/modules/2.4.27-2-386/build/include/linux/module.h:12, from zaptel.c:45: /lib/modules/2.4.27-2-386/build/include/asm/system.h: In function `__set_64bit_var': /lib/modules/2.4.27-2-386/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules /lib/modules/2.4.27-2-386/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zaptel.c: In function `zt_ctl_ioctl': zaptel.c:3476: error: `ZAPTEL_VERSION' undeclared (first use in this function) zaptel.c:3476: error: (Each undeclared identifier is reported only once zaptel.c:3476: error: for each function it appears in.) zaptel.c: In function `zt_init': zaptel.c:6838: error: `ZAPTEL_VERSION' undeclared (first use in this function) make: *** [zaptel.o] Error 1 Any ideas what I should try next? Thanks! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP hardphones with xml/html/xhtml/microbrowser support?
-BEGIN PGP SIGNED MESSAGE- Hash: RIPEMD160 right!!! it is an unfortunate description...it does not have anything to do with xml applications... Hirosh [EMAIL PROTECTED] wrote: | On Thu, 19 Jan 2006, Hirosh Dabui wrote: | | [EMAIL PROTECTED] wrote: | On Wed, 18 Jan 2006, Hirosh Dabui | wrote: | look there http://snom.com/wiki/index.php/Xmlobjects | for snom | 360... | nice... any hope for snom 320? | -Dan | ___ i think not, coz | it makes no sense on a small display... Hirosh | | | Why did snom put an xml button on the 320? (snom 320 manual page | 17, xml add-on (planned)) False hopes? :-( | | -Dan ___ --Bandwidth | and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | - -- snom technology AG Dipl.-Ing. Hirosh Dabui PGP Key-ID: 0x30A34758 mailto:[EMAIL PROTECTED] http://snom.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) iD8DBQFD1L1XAO47/DCjR1gRAymhAJsGxmMDo7ORSQ2TP8B1e/QqOMFu9wCgqYdd rxZvCeXFxJKRy8SeUBZ/Au0= =hYAs -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gen. Question
To reply to your rant the reason why I got this account is because I am out of the US (in the us for lists i used asterisk AT Dovid DOT net for the biz list asteriskusers AT dovid DOT net for users list etc). Here where I am I only have access to yahoo. I needed a name that I could remember and would not forget (I tend to forget the names to accounts that I create). So I figured I would make it asteriskdigium so I would not forget it. Anyone on this list would know that I do not work for asterisk or digium. In fact I tried to get just [EMAIL PROTECTED] but it was already taken. --- Alexander Lopez [EMAIL PROTECTED] wrote: RANT Funny your concerned about copyrights and moral issues regarding the work of others. One question you may want to ask YOURSELF is: Why would I use as my email a copyrighted work followed by the name of the Company that owns the copyright??? [EMAIL PROTECTED], Come on!! Who are you trying to fool? Are you out for the fast buck, by having someone that thinks you work for Digium hire you??? This is the same as using [EMAIL PROTECTED], or [EMAIL PROTECTED] Many people on the list may be able to look past the email as see that you are an: 1 individual that understands asterisk and digium hardware 2 just happens to have an email account at Yahoo. But many will not. If you are only using the email for the list you should have used an email that reflects that, Not an email that would server only to confuse. I would be more concerned with the bad 'vibe' of your funky email address than anything else. /rant Goong back to your question, I would add at the top of stuff you 'save' on your server, the original URL of the site it was found. That why you can give proper credit to the source. Many sites, disappear for various reasons, that's why I have a 'cache' of those sites. Voip-info being the first. I try to keep the diretory structure I get from wget so I know where it came from. [EMAIL PROTECTED] once wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Sunday, January 22, 2006 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Gen. Question Hello List, I have more of a generic question. A lot of times when links to books, little bits of codes, diffrent programs etc. are posted I do a wget to my server so I can have it for future yes. Every now and then I reply to questions with links to these kinds of things. I have never posted the URL to my server since I dont know if the one who made it would be happy giving out my link and not thiers. It's easier for me to give out the URL to my server because I tend to know what directory it is in off the top of my head. So my question basicly is, is it ok to post links for files etc. on my server that orig. came from a diffrent site ? Regards, Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP response 300 Multiple choice ???
[Jan 23 19:56:44] -- Got SIP response 300 Multiple choice back from 194.120.0.201 [Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to 'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED]' (thanks to SIP/voipstunt-5c8c) [Jan 23 19:56:44] NOTICE[3439]: chan_local.c:483 local_alloc: No such extension/context 194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED] creating local channel [Jan 23 19:56:44] NOTICE[3439]: app_dial.c:473 wait_for_answer: Unable to create local channel for call forward to 'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED]' (cause = 0) Just curious, what does it mean? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, NAT and Firewalls
As a general rule if the phone is behind NAT there should be no issues. Server behind NAT = Lots of issues (which can all be worked out). You will have to specify NAT=YES in the dial plan. Regards, Dovid --- Moises Silva [EMAIL PROTECTED] wrote: you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michaël Gaudette [EMAIL PROTECTED] wrote: Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Extensions
Hi, I am new to asteriks, ON Fedora Core 4 I installed asteriks with Asteriks Management Portal İt seems working. In the wen configuration panel i added a user and it seems added but i can not connect from sjphone soft phone. On the other way, I start asteriks console and run sip show users No users listed as i restarted asteriks. Can anyone help? Toygun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] macro-faxreceive
Technical Support wrote:
Check out www.generationd.com for their fax2mail and mail2fax scripts. It
might make life simpler
There is no description how to set-up!
The scripts are working with asterisk, but how?
bye
Ronald Wiplinger
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, January 22, 2006 9:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] macro-faxreceive
How should be the macro rewritten?
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,Set([EMAIL PROTECTED])
exten = s,104,Goto(3)
...
[Jan 23 10:43:38] -- Executing Macro(Zap/3-1, faxreceive) in new
stack
[Jan 23 10:43:38] -- Executing Set(Zap/3-1,
FAXFILE=/var/spool/asterisk-fax/1137984210.35.tif) in new stack [Jan 23
10:43:38] WARNING[25524]: pbx.c:1555 pbx_extension_helper: No application
'DBGet' for extension (macro-faxreceive, s, 2)
[Jan 23 10:43:38] == Spawn extension (macro-faxreceive, s, 2) exited
non-zero on 'Zap/3-1' in macro 'faxreceive'
[Jan 23 10:43:38] == Spawn extension (fax, 2201, 1) exited non-zero on
'Zap/3-1'
[Jan 23 10:43:38] -- Executing System(Zap/3-1,
/usr/local/sbin/mailfax /var/spool/asterisk-fax/1137984210.35.tif
) in new stack
[Jan 23 10:43:38] -- Hungup 'Zap/3-1'
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Re: [Asterisk-Users] Dundi Examples
Please stop plugging the book. Its annoying. We know its out there. http://asteriskdocs.org deserves all mentions it receives and the people behind it like Leif have done a great service to the community. The entire book is still available online free so why stop plugging it. For two years, the site online readable/downloadable PDF version was the only decent general doc for asterisk available on the planet. A huge amount of work went into this volume and if people find errors in it, they should indeed send them either to O'Reilly or Leif or Jim or one of the others involved. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom FW
I don't think you can beat the Polycom's for design, features, configuration options and functionality tho. :) Polycoms (I only have experience with a ip500) have many qualities. However, I think it's only a matter of time before entries at the $180-$200 price point begin beating it in many ways. Configuration? Yes, there are a zillion options, and many need to be hand edited in XML. The web config interface is next to useless. The Cisco/Linksys/Sipura already beats the Polycoms to a pulp on this issue. The menuing system badly needs to be redone by someone who uses a phone every day. Way too many levels on common operations. The physical design is superior, yes. The software design, sorry, no. Audio quality is the high end reference, but lower end phones come close. Functionality? Nah, a lot of cheaper phones have more and easier to use fonctionality. Again, the Sipura comes to mind. My IP500 has had more spontaneous reboots than any phone I've ever owned (many IAX2 phones do this from time to time bu they are all using beta firmware). reminderThis is an opinion and should be thought of as carved in stone. It may be of interest to people who have never seen any of these phones though./reminder IOW - YMMV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debug with ser
hi how can i debug with ser and use log() command in SER? where it will log ? thanks -- Giti Data products Trading Company Mob : +971 508715610 Tel : +971 4 2973961 Fax : +971 4 2976404 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll
This has been discussed before. The decision was that manager in Asterisk should *not* be XML. That's why we started to create the AstManproxy that converts to XML. I do believe that an XML formatted manager will help a lot of developers, so having the option of both is a good thing. Before that even happens, we need to continue cleaning up manager. There is a lot of syntax errors, re-used headers and other issues that needs to be fixed. And a lot of things that is not implemented in manager, but only in CLI with stripped output to fit a console screen. Please help us fix that first. Regards, /Olle (Who wrote two patches to convert SIPpeers and IAXpeers to xml as an option, and got stopped). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/Fstr aw poll
This has been discussed before. The decision was that manager in Asterisk should *not* be XML. That's why we started to create the AstManproxy that converts to XML. I do believe that an XML formatted manager will help a lot of developers, so having the option of both is a good thing. Before that even happens, we need to continue cleaning up manager. There is a lot of syntax errors, re-used headers and other issues that needs to be fixed. And a lot of things that is not implemented in manager, but only in CLI with stripped output to fit a console screen. Please help us fix that first. Regards, /Olle (Who wrote two patches to convert SIPpeers and IAXpeers to xml as an option, and got stopped). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is Native MoH and how do we user it
What is Native MoH, what file formats it has and how we use them. Is it SLN files or GSM? How we enable Native MoH? I've tried everything but my MP3 MoH is not going to work (very distorted). GSM voice prompts play ok over the phones. I converted fpm-calm-river.mp3 etc to GSM using sox, but output files sound terrible. And still they played distorted. But other gsm voice prompts played ok when played as MoH. What should I do now for MoH. I have everything working perfect except MoH. Why only music files are distorted and not Allison's voice prompts. Thanks in advance for help. Zach ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over using CHANAVAIL
On Sunday 22 January 2006 14:11, Chris Mason wrote:
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but this returns the provider but not the username, so I
don't understand how to use this for real applications to determine IAX2
availability. The only way I can see to use it is to only specify one
channel and test it, jumping to n+101 if it isn't.
That is pretty much how I do things. I use qualify for my SIP and IAX2
connections and then basically do something like this:
In my nufone-dial macro():
exten = s,n,Dial(IAX2/[EMAIL PROTECTED]/${ARG1},,go)
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(asterlink-dial,${ARG1},${ARG2})
And then the asterlink-dial macro is almost identical, except CHANUNAVAIL
calls the pri-dial macro, which I use as a last-effort attempt to get a call
out, as it's my most expensive route.
-A.
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Re: [Asterisk-Users] What is Native MoH and how do we user it
Its ulaw. sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql is one way to get there. You could also take a look at format_mp3 in asterisk-addons which is what I use. Chris - Original Message - From: Zach A [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 2:00 PM Subject: [Asterisk-Users] What is Native MoH and how do we user it What is Native MoH, what file formats it has and how we use them. Is it SLN files or GSM? How we enable Native MoH? I've tried everything but my MP3 MoH is not going to work (very distorted). GSM voice prompts play ok over the phones. I converted fpm-calm-river.mp3 etc to GSM using sox, but output files sound terrible. And still they played distorted. But other gsm voice prompts played ok when played as MoH. What should I do now for MoH. I have everything working perfect except MoH. Why only music files are distorted and not Allison's voice prompts. Thanks in advance for help. Zach ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing List (JUST A TEST)
Sorry, I haven't received a message in a few hours, just testing to see if it is alive. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] openH323 from cvs
Hi all, Despite of www.openh323.org and some other sites claim the cvs has an empty password for anonymous, I am unable to download the code from it. Any clue? Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/openh323 CVS password: cvs [login aborted]: reading from server: Connection reset by peer Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/openh323 CVS password: cvs [login aborted]: unrecognized auth response from cvs.sourceforge.net: M -!- Client or Server ti meout occurred! Regards, Victor. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to have a phone ring another extension as soon as off-hook?
On Fri, Jan 20, 2006 at 12:32:32PM -0500, Script Head wrote: I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? Don't know about asterisk, but some phones have that feature (Atcom AT-320). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Polycom FW
We conducted focus groups, looking at several different vendors, before we decided to go with the Polycom. From the user interface perspective, the Polycom's won hands down. I was never involved with it, but apparently to configure the Cisco's you need to be converting hex??? Yuk! -Original Message- From: Wilson Pickett [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Polycom FW I don't think you can beat the Polycom's for design, features, configuration options and functionality tho. :) Polycoms (I only have experience with a ip500) have many qualities. However, I think it's only a matter of time before entries at the $180-$200 price point begin beating it in many ways. Configuration? Yes, there are a zillion options, and many need to be hand edited in XML. The web config interface is next to useless. The Cisco/Linksys/Sipura already beats the Polycoms to a pulp on this issue. The menuing system badly needs to be redone by someone who uses a phone every day. Way too many levels on common operations. The physical design is superior, yes. The software design, sorry, no. Audio quality is the high end reference, but lower end phones come close. Functionality? Nah, a lot of cheaper phones have more and easier to use fonctionality. Again, the Sipura comes to mind. My IP500 has had more spontaneous reboots than any phone I've ever owned (many IAX2 phones do this from time to time bu they are all using beta firmware). reminderThis is an opinion and should be thought of as carved in stone. It may be of interest to people who have never seen any of these phones though./reminder IOW - YMMV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] debug with ser
Example: log (L_INFO,test) It will go to syslog, ie /var/log/messages. Douglas. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 6:48 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] debug with ser hi how can i debug with ser and use log() command in SER? where it will log ? thanks -- Giti Data products Trading Company Mob : +971 508715610 Tel : +971 4 2973961 Fax : +971 4 2976404 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom FW
Douglas Garstang wrote: We conducted focus groups, looking at several different vendors, before we decided to go with the Polycom. From the user interface perspective, the Polycom's won hands down. I was never involved with it, but apparently to configure the Cisco's you need to be converting hex??? Yuk! This is not correct. The Polycom and Cisco phone configuration is very similar. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
Greg Boehnlein wrote: (Steve Totaro wrote:) What I would really like to do is have one D channel coming in on the T3 and have it split between each of the T1/PRI or even better one D channel per quad (I know Asterisk can do that). Is it possible? No. Actually, it is, using an Adtran Atlas with a DS3 interface and DS1 interfaces. Not cheap, but possible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
Greg Boehnlein wrote: Hehehe.. Ask your Telco if they can provision E1 for you. ;) The Digium cards can handle E1 or T1, and if you go E1 you'll get 30 channels instead of 24 on the span. I have talked to a number of telcos in the US about this... they don't have the ability to do it. Ignoring the NI-2 vs. EuroISDN issue (Asterisk can easily run NI-2 over an E1, some switches cannot), their networks _cannot_ handle a 2.048MHz span. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
Greg Oliver wrote: I am unsure of * capabilities on NFAS (we do not use PCs to terminate any PRIs), but it allows bonding of desparate PRIs to use a single d-channel. ie, you can have 1 d-channel (optional backups) for the entire DS3. Not sure if * can communicate across cards like that in the same bus though. At the moment Asterisk cannot do NFAS across multiple servers, but Matt F and I have been discussing a possible method for doing it. Don't be surprised if it shows up in the development branch in the near future :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi Examples
Wilson Pickett wrote: Please stop plugging the book. Its annoying. We know its out there. http://asteriskdocs.org deserves all mentions it receives and the people behind it like Leif have done a great service to the community. The entire book is still available online free so why stop plugging it. For two years, the site online readable/downloadable PDF version was the only decent general doc for asterisk available on the planet. A huge amount of work went into this volume and if people find errors in it, they should indeed send them either to O'Reilly or Leif or Jim or one of the others involved. ___ You left the attribution off the quote you included with your mail. That was Dovid Bender, right? Thx. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bug in attended transfer or as expected?
The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. If you have enabled Disconnect Call feature, then you can hangup with *0 for example, that will hangup only the current call, not the call on hold. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom FW
Doug Lytle wrote: Douglas Garstang wrote: We conducted focus groups, looking at several different vendors, before we decided to go with the Polycom. From the user interface perspective, the Polycom's won hands down. I was never involved with it, but apparently to configure the Cisco's you need to be converting hex??? Yuk! This is not correct. The Polycom and Cisco phone configuration is very similar. Does anyone know whether the reports of the errors in the Asterisk book wrt to Dundi were correct or not? Anytime I read a technical posting that is written with such a harsh tone, I wonder if it has any meat to it. . . B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail over using CHANAVAIL
I am trying to construct a macro for long distance dialling. I have
two internet feeds, I have all routes including Teliax on
Internet A
and a static route to Voxee on Internet B.
Here's an AEL macro I use on our boxes. Modify for your needs.
// dial a number with a range of routing options
macro outbound (number, clid, route1, route2, route3, route4) {
if (${clid} = ) {
CALLERID(number)=${DEFAULTCID};
} else
CALLERID(number)=${clid};
dialstart:
switch (${route1}) {
case direct:
dialout (${number});
break;
case providera:
dialout (IAX2/providera/${number});
break;
case providerb:
dialout (IAX2/providerb/${number});
break;
case providerc:
dialout (SIP/[EMAIL PROTECTED]);
break;
case pstn:
dialout (SIP/[EMAIL PROTECTED]);
break;
default:
NoOp (invalid route: ${route1});
};
route1=${route2};
route2=${route3};
route3=${route4};
if (${route1} = ) {
Playtones (info);
Congestion ();
};
goto dialstart;
};
// dial a number ignoring anything except busy
macro dialout (dialstring) {
Dial (${dialstring},,TW);
switch (${DIALSTATUS}) {
case BUSY:
Playtones (busy);
Busy ();
break;
};
};
Basically, replace dial commands in extensions.conf with a call to macro
outbound, passing it the number to dial, callerid to present, and any
number of routes in the order you want them to be tried. The macro dialout
just ensures that if the number called is genuinely busy, outbound doesn't
plough on with routes 2,3,4 regardless.
Hope that helps.
Regards,
Chris
--
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[Asterisk-Users] Re: Polycom boot times/XML files.
Andrew Furey wrote: Huh? My 7905 takes well under 10 seconds, including Asterisk registration and NTP update. Granted, if it were DHCP it might take marginally longer, but 5 _minutes_? Yeah, the Polycoms *do* take a while to boot -- but not five minutes. I've timed mine (Polycom 501's) and it's 1:25. Ain't exactly zippy, but for something that only reboots when I'm servicing it, it's acceptable. Aside from that, I really dig the Polycom phones: good looks, good audio, good menuing. [Note: the x01's are a fair improvement over the x00's, which took a smidge longer to boot, and had not-as-nice menus and overall look-and-feel.] As for someone complaining that you had to hand-edit the XML files, instead of using a nice GUI... guess what: that's what scripting is for. I can tell you that I'll spend 15 minutes to write a script, and be able to spawn it onto 50 phones in ten seconds ANY DAY over having to fire up a browser to those same 50 phones. And God help you if you had a real number of phones to change. For that matter, some day I'll get off my lazy a** and write something that talks to the dhcpd.leases file and auto-provisions new phones. -Ken Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID
I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten = 1102,1,Answer()
exten = 1102,2,Set(DIALEDNUM=1102)
exten = 1102,3,Wait(2)
exten = 1102,4,Playback(pls-wait-connect-call)
exten = 1102,5,Wait(1)
exten = 1102,6,Dial(SIP/2102SIP/3102SIP/4102SIP/[EMAIL PROTECTED],
33,mj)
exten = 1102,7,Voicemail(su{$EXTEN})
exten = 1102,8,Hangup()
exten = 1102,106,Voicemail(sb{$EXTEN})
exten = 1102,107,Hangup()
where porta is my SIP account with the company that provides my
PSTN connection. I know for a fact that I can set any caller ID I
want (because I've done it with ATAs) and my carrier will pass it;
however, my question is, how do I get my asterisk box to pass the
original Call*ID instead of the number assigned to me by my provider?
this is the entry in sip.conf
[porta]
type=peer
secret=corbe9845
username=portasip
host=68.145.125.95
;fromuser=17862065496
fromdomain=66.165.175.35
insecure=very
;nat=yes
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Re: [Asterisk-Users] T3 Mux and Asterisk Question
Kevin P. Fleming wrote: Greg Boehnlein wrote: Hehehe.. Ask your Telco if they can provision E1 for you. ;) The Digium cards can handle E1 or T1, and if you go E1 you'll get 30 channels instead of 24 on the span. I have talked to a number of telcos in the US about this... they don't have the ability to do it. Ignoring the NI-2 vs. EuroISDN issue (Asterisk can easily run NI-2 over an E1, some switches cannot), their networks _cannot_ handle a 2.048MHz span. Actually every US made switch I've ever seen is 2.048MHz to the core. They then rate change to the T1s. It makes export easier to handle. Any mixof E1/2/3.. or T1/2/3.. cards will just plug in and go timing wise. The issue is probably more of not being set up for mixed ulaw/Alaw working. I don't think anyone is really set up for that within the switch. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T3 Mux and Asterisk Question
Steve Underwood wrote: Actually every US made switch I've ever seen is 2.048MHz to the core. They then rate change to the T1s. It makes export easier to handle. Any mixof E1/2/3.. or T1/2/3.. cards will just plug in and go timing wise. The issue is probably more of not being set up for mixed ulaw/Alaw working. I don't think anyone is really set up for that within the switch. It's not the switches, it's the DACS/mux/SONET networks they are attached to for span delivery to the customers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] not able to start asterisk
Hi
iam not able to start asterisk
give me following error
any help
STARTING ASTERISK/usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}Asterisk ended with exit status 132
Asterisk exited on signal 4.Automatically restarting Asterisk./usr/sbin/safe_asterisk: line 42: 4637 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132Asterisk exited on signal 4.Automatically restarting Asterisk.mpg123: no process killed
-Asterisk could not start!Use 'tail /var/log/asterisk/full' to find out why.-
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[Asterisk-Users] Re: Asterisk Development and Release Cycle
OK, so if I were using SVN, the stable branch would still be changing and my problem was that I was using the Tarballs? Correct? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Steven [EMAIL PROTECTED] wrote: This is great news. Agreed! Previously, stable was just considered a snapshot and if you ran stable and encountered a bug, you had to switch to head to get the fix. I don't think this is correct. Pure bug fixes were always applied to the stable 1.0 branch. Where you needed to use head was if you wanted to use the cool new features that were never going into 1.0. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] not able to start asterisk
My first guess would be to
Use 'tail /var/log/asterisk/full' to find out why.
On 1/23/06, ram [EMAIL PROTECTED] wrote:
Hi
iam not able to start asterisk
give me following error
any help
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core
dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 42: 4637 Illegal instruction (core
dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
mpg123: no process killed
-
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-
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Re: [Asterisk-Users] Music on Hold
What I did so far: My [EMAIL PROTECTED] PBX is working fine, with four Grandstream Budge Tone –100 phones. Regarding the MoH feature, I did the following: Checked the presence of mpg123. It is. 2. In /etc/asterisk/zapata.conf, I added the line "musiconhold=default" under [channels] context In /etc/asterisk/musiconhold.conf, I uncommented the line that says "default = mp3:/var/lib/asterisk/mohmp3”. I restarted Asterisk in order to reload the musiconhold.conf settings My files are all mp3 (no ID3 tags left), and I used the AMP Portal, Setup, On Hold Music, and replaced the default files with some new ones. Please help Thanks, Ed Zaldibar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Announcing PodMail 1.0 (GPL)
Hello Asterisk Community. While sitting at lunch the other day I had a typical napkin-prototype idea: What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0 PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting. PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes. PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible. Live Demo: Do not miss out our live demo at http://podmail.alkaloid.net/ Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately! Check out the PodMail Documentation and Installation Notes at http://projects.alkaloid.net. PodMail is released under the terms of the GPL. Enjoy! /BAK/ -- Ben Klang Alkaloid Networks http://projects.alkaloid.net [EMAIL PROTECTED] 404.475.4850 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi Examples
At 05:06 AM 01/23/2006, you wrote: http://asteriskdocs.org deserves all mentions it receives and the Though you really should mention that it's a 1.0 document and trying to make a 1.2 installation work using that book is somewhat futile. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 and Cisco IOS 12.4
I have the same issue. I just bought the commercial version from Digium to see if that has the same problem. I wanted to use the free one to test out g729. My Polycom 301 had no issues using the free codec though (testing via VM, etc) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Weiser Sent: Tuesday, December 20, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] G729 and Cisco IOS 12.4 Can anyone confirm that when using the G729 codec from http://kvin.lv/ pub/Linux/Asterisk/ and a Cisco gateway running IOS 12.4, codec negotiation fails? When I configure the dial-peer in the router with g729r8, it fails. If I use g729br8 (which uses a built-in VAD), it works. This behavior started since we upgraded the router from 12.3 (which had no issues). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP over TCP: latest news?
Hi, I know it is a FAQ but I'm interested in latest news (if any...) about SIP over TCP support in Asterisk. I found this: https://savannah.nongnu.org/projects/asterisk-tcp/ but I'm not able to understand if project is active and what is its level of development. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], Steven [EMAIL PROTECTED] wrote: OK, so if I were using SVN, the stable branch would still be changing and my problem was that I was using the Tarballs? Correct? I guess so. The stable branch (now branches/1.2) has bug fixes applied to it as they get done. Every so often the current state of the branch will get tagged 1.2.x and released as a tarball. Cheers Tony -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Tony Mountifield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] In article [EMAIL PROTECTED], Steven [EMAIL PROTECTED] wrote: This is great news. Agreed! Previously, stable was just considered a snapshot and if you ran stable and encountered a bug, you had to switch to head to get the fix. I don't think this is correct. Pure bug fixes were always applied to the stable 1.0 branch. Where you needed to use head was if you wanted to use the cool new features that were never going into 1.0. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729a Pass-Through and Recording/Monitoring
Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a big performance hit in this setup whether or not a license is required? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
On 1/23/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote: On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote: What is the problem with step 3? See this example as basis for modifications: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro Unless I have terribly misunderstood that macro, that is basically the same thing I am doing now, is it not? Simply transfer the customer to a conference room (I might have a look into the automatically determined conf room number), then transfer all collegues in there as well and finally jump in myself. It is however not quite what I described in step 3. Yes, that's so. I tested this macro with SIP-softphones and it works. May be this is the simplest way to do what you want. And this is a good start point for modifications. -- Good luck, Alexander Chemeris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 and Cisco IOS 12.4
Same thing...even with the commercial Digium G729 codec. I have to specifiy G729br8 on the Cisco. Cisco issue? Bill -Original Message- From: Bill Gibbs Sent: Monday, January 23, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] G729 and Cisco IOS 12.4 I have the same issue. I just bought the commercial version from Digium to see if that has the same problem. I wanted to use the free one to test out g729. My Polycom 301 had no issues using the free codec though (testing via VM, etc) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Weiser Sent: Tuesday, December 20, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] G729 and Cisco IOS 12.4 Can anyone confirm that when using the G729 codec from http://kvin.lv/ pub/Linux/Asterisk/ and a Cisco gateway running IOS 12.4, codec negotiation fails? When I configure the dial-peer in the router with g729r8, it fails. If I use g729br8 (which uses a built-in VAD), it works. This behavior started since we upgraded the router from 12.3 (which had no issues). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320 and message retrieve key
On 08:08, Mon 23 Jan 06, Alex Barnes wrote:
I expect the issue is the same problem we have with the 360's.
Quick fix is add the old Snom MWI fix to your dial plan but its not perfect
solution for us
as all our phones with DDI present 6 digits and we have already created our
mailboxes to match
the 3 digit ext number which means the users have to enter their mailbox
number as well as password.
Then use this:
VoicemailMain(${CALLERIDNUM:3})
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] not able to start asterisk
there is no File called that name in that place
that is the reason i have mailed here
ram
On 1/23/06, John Broome [EMAIL PROTECTED] wrote:
My first guess would be toUse 'tail /var/log/asterisk/full' to find out why.On 1/23/06, ram
[EMAIL PROTECTED] wrote: Hi iam not able to start asterisk give me following error any help STARTING ASTERISK
/usr/sbin/safe_asterisk: line 42:4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY} Asterisk ended with exit status 132
Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42:4637 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY}
/dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. mpg123: no process killed -
Asterisk could not start! Use 'tail /var/log/asterisk/full' to find out why. - ___ --Bandwidth and Colocation provided by
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[Asterisk-Users] Problem with Codecs
Hi All, I configured Asterisk and it is working successfully with Express Talk. Now I am trying to work with some other client which supports only GSM and now Asterisk never worked and tried to make a call out. In sip.conf I disallowed all and allowed only GSM also. I also heard that Asterisk does transcoding automatically and I have no clue where should I change my configuration. Someone please help me to make Asterisk work with GSM. Thanks, Manoj. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firewall/Embeded System/CF/etc
I am trying to build an silent non moving parts (fans,HD.etc) embedded system...Firewall/Asterisk/FXo/FXs/CF/etc Looking for anyone running asterisk with Coyote, IPcop, m0n0wal, Shorewall, etc in the same system/box!!! Offlist please... Thanks in advance!! Manny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom videoconferencing with asterisk?
Hello, Has anyone used Polycom's VSX line of videoconferencing equipment with Asterisk? It seems some of their models, namely the newer VSX 5000, supports SIP. -- The Internet used to be a lot of smart people sitting at dumb terminals, but now its a lot of dumb people sitting at smart terminals! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Teliax Down?
Ive had problems for the last couple of weeks regarding incoming calls. Cant hear the party calling me (their voice sounds garbled/scrambled). If you havent done so yet, I would recommend you post your complaint on their online forum as well under bugs. You usually get some good responses from other Teliax users regarding the problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ross C Sent: Friday, January 20, 2006 8:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Down? I was having trouble too. I had trouble yesterday as well. I called and David said it was a massive DDOS. Seems to get fixed pretty quickly when it does happen (5 minutes or so); however, for a business, 5 minutes without phones (people cant get a hold of your company) isnt really acceptable IMO. Also on co3. I couldnt even access their website during that time From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty Dekema Sent: Friday, January 20, 2006 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Teliax Down? Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; voip-co3.teliax.com. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Home Test!
Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing List (JUST A TEST)
we hear you loud and clear 2006/1/23, [EMAIL PROTECTED] [EMAIL PROTECTED]: Sorry, I haven't received a message in a few hours, just testing to see if it is alive. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to have a phone ring another extension as soon as off-hook?
On 01/23/2006 08:55:09 AM, Omadon wrote: On Fri, Jan 20, 2006 at 12:32:32PM -0500, Script Head wrote: I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? I believe you would do this by having extension 's' dial the second phone. I believe it would be best to have a new extension context for this code. Then, you need to associate your new context in extentionss.conf with the channel your phone uses. Anyhow, I think this will work with a zapta interface and a plain old telephone. I don't know if the fancy network phones try to contact asterisk/anybody when the receiver is picked up. Karl [EMAIL PROTECTED] Free Software: You don't pay back, you pay forward. -- Robert A. Heinlein ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Down?
I hate to burst your bubble but DOS attacks are a fact of life for IP based services. The bigger you get the more of a target you are. There are a ton of DOS prevention/mitigation appliances/services available in today's world. But they all rely on the same thing: having more bandwidth/capacity than your attacker. I've seen DOS attacks against ISP customers of mine that were pushing over a million packets per second across 50+ peering points. Not many networks can absorb that kind of thing. If your phones are that critical to your business you need to get dedicated service (aka T1), or switch to a service with static registration that can be protected with a good firewall. Max On 1/23/06, JCC [EMAIL PROTECTED] wrote: I've had problems for the last couple of weeks regarding incoming calls. Cant hear the party calling me (their voice sounds garbled/scrambled). If you haven't done so yet, I would recommend you post your complaint on their online forum as well under 'bugs'. You usually get some good responses from other Teliax users regarding the problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Ross CSent: Friday, January 20, 2006 8:40 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Teliax Down? I was having trouble too. I had trouble yesterday as well. I called and David said it was a "massive DDOS". Seems to get fixed pretty quickly when it does happen (5 minutes or so); however, for a business, 5 minutes without phones (people can't get a hold of your company) isn't really acceptable IMO. Also on co3. I couldn't even access their website during that time… From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Rusty DekemaSent: Friday, January 20, 2006 5:42 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Teliax Down? Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; voip-co3.teliax.com. -Rusty___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Max Clarkhttp://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to view Q.931 Disconnect code
Hi there,Can anyone know how to view asterisk disconnect code.?-- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807 DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I've sent a message to the list 6 hours ago and it's still not showing up
There seems to be a queue of some sort the messages fit through. No matter how long it seems to take, all messages I've sent get through, even 12+ hours later. I've done my share of double-postings and have learned to wait ;) I haven't discerned anything broken with the list, just slow maybe when under load. Roger Hanson wrote: I've sent a message to the list Asterisk-user 6 hours ago and it's still not showing up. I've seen others with questions about the availability of the list. It may be something the moderators want to check out. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk for Call Center (missing reference)
Hi, Does any body knows some thing about it? Thanks in advance. Telles Rodrigo P. Telles wrote: Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from John Todd asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you found a solution for your question? if so, please let me know! Thanks in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 videoconferencing with asterisk?
Hello: I´ll like to know if asterisk is capable of making H.323 videoconferencing and if it can also transcode fromH.323 to SIP Any help will be appreciate Tanks Erick W. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd asterisk behavoir
Hi, If I have an AGI script that calls user A and then calls user B and connects them... it seems to work fine (for accounting) if I call a local call (out my PRI).. however if I go out my IAX... the CDR terminates the long distance call after 3 seconds (after the IAX trunk picks up).. and what ends up in the CDR is a time .. but it's FROM (src) the long distance call to my local extension.. which doesn't help why is it doing that? Jan 23 13:39:57 DEBUG[28521] chan_iax2.c: Ooh, voice format changed to 4 Jan 23 13:39:59 DEBUG[28521] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jan 23 13:39:59 VERBOSE[28521] logger.c: -- IAX2/calleveryone-1 is ringing Jan 23 13:39:59 VERBOSE[28521] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Jan 23 13:39:59 DEBUG[28521] chan_zap.c: Requested indication 3 on channel Zap/1-1 Jan 23 13:39:59 VERBOSE[28521] logger.c: -- IAX2/calleveryone-1 stopped sounds Jan 23 13:39:59 VERBOSE[28521] logger.c: -- Local/[EMAIL PROTECTED],1 stopped sounds Jan 23 13:39:59 DEBUG[28521] chan_zap.c: Requested indication -1 on channel Zap/1-1 Jan 23 13:39:59 VERBOSE[28521] logger.c: -- IAX2/calleveryone-1 answered Local/[EMAIL PROTECTED],2 Jan 23 13:39:59 VERBOSE[28521] logger.c: -- Local/[EMAIL PROTECTED],1 answered Zap/1-1 Jan 23 13:39:59 DEBUG[28521] channel.c: Planning to masquerade channel IAX2/calleveryone-1 into the structure of Local/[EMAIL PROTECTED],1 Jan 23 13:39:59 DEBUG[28521] channel.c: Done planning to masquerade channel IAX2/calleveryone-1 into the structure of Local/[EMAIL PROTECTED],1 Jan 23 13:39:59 DEBUG[28521] chan_local.c: Not posting to queue since already masked on 'Local/[EMAIL PROTECTED],2' Jan 23 13:39:59 DEBUG[28521] channel.c: Got clone lock for masquerade on 'IAX2/calleveryone-1' at 0x8db3a64 Jan 23 13:39:59 DEBUG[28521] channel.c: Putting channel IAX2/calleveryone-1 in 64/64 formats Jan 23 13:39:59 DEBUG[28521] channel.c: Released clone lock on 'Local/[EMAIL PROTECTED],1ZOMBIE' Jan 23 13:39:59 DEBUG[28521] channel.c: Done Masquerading IAX2/calleveryone-1 (6) Jan 23 13:39:59 DEBUG[28521] channel.c: Didn't get a frame from channel: Local/[EMAIL PROTECTED],2 Jan 23 13:39:59 DEBUG[28521] channel.c: Bridge stops bridging channels Local/[EMAIL PROTECTED],2 and Local/[EMAIL PROTECTED],1ZOMBIE Jan 23 13:39:59 DEBUG[28521] app_dial.c: Exiting with DIALSTATUS=ANSWER. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering Service Add-on?
Anybody seen some client/server asterisk add-on script for "live" answering services to provide call handling and message taking from an Operator? Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to view Q.931 Disconnect code
Hi,
Try
exten = h,1,NoOp(${HANGUPCAUSE})
in your extensions.conf
Cheers.
Andy
On 1/23/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi there,
Can anyone know how to view asterisk disconnect code.?
--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +63 917 542 5807
DID: (+63) 44 7906770
US DID: +1 619 399 0128
msn: [EMAIL PROTECTED]
skype: bulcrack
___
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RE: [Asterisk-Users] Answering Service Add-on?
Not sure what you mean but a basic PBX does what I have read. -Original Message- From: Bart Fisher Sent: Mon 1/23/2006 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] Answering Service Add-on? Anybody seen some client/server asterisk add-on script for live answering services to provide call handling and message taking from an Operator? Bart winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Waiting CallerID
Hi, According to the wiki, we need to have both callwaiting=yes and callwaitingcallerid=yes , and that's what I have in zapata.conf. I can hear the call waiting alert tone when a 2nd call comes in during an established call, and I can switch between the calls without problems. However, CallerID on the 2nd call does not show up with the call waithing alert tones. Am I missing something? Can anyone help? Thank you in advance. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
Yep I did the same. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, 21 January 2006 5:34 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? On Sat, January 21, 2006 23:21, Franz Bräuer said: Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? Kewl! Those work like a treat! As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: cd /usr/lib/asterisk/modules/ wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so After reloading, 'show translation' gives: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 -22 8 817 8 724 115 19897 gsm 151 - 7 716 7 623 114 19796 ulaw 14616 - 111 2 118 109 19291 alaw 14616 1 -11 2 118 109 19291 g726 154241010 -10 926 117 20099 adpcm 14616 2 211 - 118 109 19291 slin 14515 1 110 1 -17 108 19190 lpc10 161311717261716 - 124 207 106 g729 16939252534252441 - 215 114 speex 16030161625161532 123 - 105 ilbc 17343292938292845 136 219 - Jolly good show, old chap! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video Conferencing.
I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial out and message playback
look at this: http://www.voip-info.org/wiki-VICIDIAL+Dialer perhaps it's what you are looking for... 2006/1/23, Danish Samad [EMAIL PROTECTED]: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Recording
The option for MeetMe() to record(r) the conference does not seem to be working. I see a CLI message that it is starting recording, however no file is ever created. No error or warnings messages are seen either. Starting recording of MeetMe Conference 100 into file meetme-conf-rec-100-1138045561.0.wav. extension = 100,MeetMe(,r) Is there something that I am missing to get this to work? --johann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
Erick Muchas Gracias por la respuesta. I'm not using any of that projects, it's my own Asterisk installation onto slackware 10. well what can you tell about sipura ones? 2006/1/23, Erick Perez [EMAIL PROTECTED]: Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial out and message playback
Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial out and message playback
An example of this would be Outcall Voice Mail? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Danish Samad wrote: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] user not seen
Hi, I installed Asterisk yesterden with amportal, I added 2 sip extensions, and it is seen in mysql too. But when i try to register from any device or softphones, invalid username/secret message comes, İn tne cl, i us esip show users -- no users But user is in mysql db What can i do Toygun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial out and message playback
Danish Samad wrote: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish Take a look at call files as a starting point. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video Conferencing.
It's possible to do point to point but not point to multipoint. I tried to get development for this some time ago and no one responded, check out my Video Conference Bounty on www.voip-info.org, since then we have developed our own solution that we have decided to market, it will cost $2,000 for up to 10 users that uses the Macromedia communications server. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: Monday, 23 January 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing. I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
I haven't worked with sipura. So I can't write about it. If I stick to the reviews, then it is a good/stable product with some minor/strange/rarely-ocurred issues regarding phantom calls. spanish-onno creas que no hablo español, pero sabes que aqui solo puedes postear en ingles no?spanish-off On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Erick Muchas Gracias por la respuesta. I'm not using any of that projects, it's my own Asterisk installation onto slackware 10. well what can you tell about sipura ones? 2006/1/23, Erick Perez [EMAIL PROTECTED]: Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_rxfax.so and app_txfax.so
Hi, I search in the archives and I don't find that case. I'm wanted todo Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler Ifsomebody can help me it would be appreciate, Loic Foucault ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] dial out and message playback
On Tue, 24 Jan 2006, Danish Samad wrote: Hi, -users questions In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? use .call files in /var/spool/asterisk/outgoing I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish - wasim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 - the party's over :-(
The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Buy them while you still can :-( -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
Cute? But it can use LDAP... PaulH - Original Message - From: Ben Klang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 3:58 AM Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Hello Asterisk Community. While sitting at lunch the other day I had a typical napkin-prototype idea: What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0 PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting. PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes. PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible. Live Demo: Do not miss out our live demo at http://podmail.alkaloid.net/ Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately! Check out the PodMail Documentation and Installation Notes at http://projects.alkaloid.net. PodMail is released under the terms of the GPL. Enjoy! /BAK/ -- Ben Klang Alkaloid Networks http://projects.alkaloid.net [EMAIL PROTECTED] 404.475.4850 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] background SayDigits()?
Is it possible to background SayDigits()?
I know you can manually Background() each digit individually, but this
does not solve the problem when you need to do something like
SayDigits(${EXTEN}) or SayDigits(${CALLERID(number)})
-Dan
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RE: [Asterisk-Users] Home Test!
We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_rxfax.so and app_txfax.so
Support Internet.net wrote: I'm wanted to do Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler Just a guess, You've installed more then one version of spandsp. Remove all modules and libraries and re-install spandsp. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: setting outgoing caller ID by the queue an extension is logged into
Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, welog inby the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_rxfax.so and app_txfax.so
What version of SpanDSP are you running? You should be running -pre21 -Original Message-From: Support Internet.net [mailto:[EMAIL PROTECTED]Sent: Monday, January 23, 2006 1:18 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] app_rxfax.so and app_txfax.so Hi, I search in the archives and I don't find that case. I'm wanted todo Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler Ifsomebody can help me it would be appreciate, Loic Foucault ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0 SIP Trunk
Hi,I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each, the alternative is cisco 7940s ( around $250 ) running SCCP through the CCM. at the quantities I'm talking about, $100 is significant.Does anyone have any idea how to get this done? I've tried this:exten = 123,1,Dial(SIP/sipphone,20)exten = 123,2,Dial(SIP/ccm/3040)where 3040 is our VM pilot for ccm. but all it does is take us to the main greeting.we have smartnet, but they haven't been helpful at all I called digium to see if they could help if we paid, but they said they've never heard of cisco unityhelp?thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
