Re: [asterisk-users] Requires
On Mon, Jul 18, 2011 at 11:02:16AM +0530, mahesh katta wrote: Sorry boss Best Regards, Mahesh, I'm afraid that at some point Ashirwad will become annoyed that you are including the asterisk-users list on these emails. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compact, affordable x86 devices?
Hello I'd like to build a compact, affordable, fanless x86 solution to handle my home landline. I know about the following two platforms: 1. www.pcengines.ch/alix.htm alix1d + case 100 Does Availability 500 mean that it's just not possible to buy just one item? 2. www.soekris.com/products.html?limit=all net4501-30 Board and Case $175.00 Is the net4501 powerful enough to run Asterisk, considering that I'll use an external VoIP gateway to connect it to my landline? Are there other manufacturers I should know about? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro issue under 1.8.5
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing
as
app_macro has been installed okay?
Macro was depreciated in 1.6 and most likely removed in 1.8.5
Removed, no. However in future version of Asterisk it will not
be
enabled in menuselect by default.
@OP: *CLI module load app_macro.so
Same problem even after performing the above load. module does
exist:
Watch the console carefully for errors when you run that command.
They
should tell you exactly what's wrong.
Also, it may help to inspect the differences in apps/app_macro.c
between
1.8.3 and 1.8.5.
Well it seems like its getting worse!
[Jul 18 11:36:00] WARNING[28936]: pbx.c:4071 pbx_extension_helper: No
application 'Playback' for extension (home, 400, 1)
Looking in pbx.c it would appear it cannot find the application in some sort of
cache:
if (!e-cached_app)
e-cached_app = pbx_findapp(e-app);
app = e-cached_app;
ast_unlock_contexts();
if (!app) {
ast_log(LOG_WARNING, No application '%s' for extension (%s, %s, %d)\n,
e-app, context, exten, priority);
return -1;
}
Any thoughts ?
--
Thanks, Phil--
_
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Re: [asterisk-users] Asterisk binaries on CentOS version 6
A J Stiles asterisk_l...@earthshod.co.uk writes: Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even take long anymore (on any target system with the grunt to run Asterisk). The only thing to beware of is, if configure complains that you need a package that you already have, then you need the corresponding -devel package. There are advantages to packaging systems though. E.g. you never end up with an outdated module causing trouble in a newer version. You can probably make your own RPM's for CentOS 6 based on either the Digium RPM's for CentOS 5 or the Fedora RPM's for Fedora 15. Just download the source RPM and rebuild it; if you don't get any errors you are generally golden. For extra points, install mock and let that do the rebuild. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Seg Faults with 1.6.2.19
Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX with SIP
Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
Gilles wrote: Does Availability500 mean that it's just not possible to buy just one item? I read it as they have over 500 in stock. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote: Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. Thanks Lee I've not had a problem here with 1.6.2.19. What are you reloading that causes the issue, and can you post the usual gdb backtrace somewhere? Perhaps on the bug tracker. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX with SIP
On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
Hi Steve, I think it's related to my ODBC connection. I started with a random hang where it looked ODBC related which led me to try a few things. Reloading the config a few times is causing core dumps which 1.6.2.18.2 just doesn't have, however my main reason for using 1.6.2.19 is a fix to ODBC so I don't really want to downgrade. I will try and get some traces from one of my test boxes. Thanks Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: 18 July 2011 12:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote: Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. Thanks Lee I've not had a problem here with 1.6.2.19. What are you reloading that causes the issue, and can you post the usual gdb backtrace somewhere? Perhaps on the bug tracker. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
Just about any of the HP thin clients, either new or used off eBay, with AstLinux installed do a wonderful job, especially if you are not going to need a PCI card. The older units will need a larger flash. Transcend has several different sizes that are direct replacements Looks like some of the Neoware units will also do the job. John Novack Gilles wrote: Hello I'd like to build a compact, affordable, fanless x86 solution to handle my home landline. I know about the following two platforms: 1. www.pcengines.ch/alix.htm alix1d + case 100€ Does Availability500 mean that it's just not possible to buy just one item? 2. www.soekris.com/products.html?limit=all net4501-30 Board and Case $175.00 Is the net4501 powerful enough to run Asterisk, considering that I'll use an external VoIP gateway to connect it to my landline? Are there other manufacturers I should know about? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
On Mon, 18 Jul 2011 08:04:31 -0400, John Novack jnov...@stromberg-carlson.org wrote: Just about any of the HP thin clients, either new or used off eBay, with AstLinux installed do a wonderful job, especially if you are not going to need a PCI card. The older units will need a larger flash. Transcend has several different sizes that are direct replacements Looks like some of the Neoware units will also do the job. Thanks for the tip. I'd like to buy the unit new: Are those devices still manufactured? How easy is it to reflash them to run as a stand-alone Linux host? Which device would you recommend to Asterisk and a couple of other apps (small web server, SQLite, etc.)? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
First they came and said that instead of offices, doors and hallways, we should have massive, open-plan seating or grungy, industrial cubicle farms, because open spaces mean open companies! It's safe to say the advice did not fall on deaf ears. Now, we're ready to take openness to the next level. Is asterisk-users ready to be copied on all internal company correspondence? Challenge accepted. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Monday, July 18, 2011 7:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Seg Faults with 1.6.2.19 Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. We experienced the same thing. After a few reloads, Asterisk crashes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
Boy if only it was Enron :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, July 18, 2011 8:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Requires First they came and said that instead of offices, doors and hallways, we should have massive, open-plan seating or grungy, industrial cubicle farms, because open spaces mean open companies! It's safe to say the advice did not fall on deaf ears. Now, we're ready to take openness to the next level. Is asterisk-users ready to be copied on all internal company correspondence? Challenge accepted. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
Hi Eric, are you using ODBC? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 18 July 2011 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Monday, July 18, 2011 7:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Seg Faults with 1.6.2.19 Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. We experienced the same thing. After a few reloads, Asterisk crashes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
HP still does make Thin Clients, often with XP Embedded, though I have had very good results with many older used ones sold on eBay. With a new larger flash from Transcend, they simply work. Consider the used units not worn out but simply ones with more experience that probably won't fail. New units are always subject to infant mortality!! The T5720 often comes with a large enough flash that replacement isn't needed, and reflashing with AstLinux can be done a number of ways, beyond the scope of this list. AstLinux has a web server for configuration, not sure about SQlite. Check out their site for more details there are other low cost solutions around as well. the ALIX boards I have seen do not impress me. I think they are somewhat overpriced. Jut one opinion John Novack Gilles wrote: On Mon, 18 Jul 2011 08:04:31 -0400, John Novack jnov...@stromberg-carlson.org wrote: Just about any of the HP thin clients, either new or used off eBay, with AstLinux installed do a wonderful job, especially if you are not going to need a PCI card. The older units will need a larger flash. Transcend has several different sizes that are direct replacements Looks like some of the Neoware units will also do the job. Thanks for the tip. I'd like to buy the unit new: Are those devices still manufactured? How easy is it to reflash them to run as a stand-alone Linux host? Which device would you recommend to Asterisk and a couple of other apps (small web server, SQLite, etc.)? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: 18 July 2011 14:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 Hi Eric, are you using ODBC? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 18 July 2011 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Monday, July 18, 2011 7:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Seg Faults with 1.6.2.19 Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. We experienced the same thing. After a few reloads, Asterisk crashes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote: Hi Steve, I think it's related to my ODBC connection. I started with a random hang where it looked ODBC related which led me to try a few things. Reloading the config a few times is causing core dumps which 1.6.2.18.2 just doesn't have, however my main reason for using 1.6.2.19 is a fix to ODBC so I don't really want to downgrade. I will try and get some traces from one of my test boxes. Thanks Lee I can confirm that we are NOT using ODBC, and that our box does NOT crash, so your theory is still holding up. :) Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
On Mon, 18 Jul 2011 09:03:52 -0400, John Novack jnov...@stromberg-carlson.org wrote: there are other low cost solutions around as well. the ALIX boards I have seen do not impress me. I think they are somewhat overpriced. Jut one opinion Thanks for the feedback. I'll read what HP has to offer. When you mention other low-cost solutions, I assume you mean other thin clients reflashed to run as stand-alone hosts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
On 07/18/2011 09:00 AM, Robert Huddleston wrote: Boy if only it was Enron :) Baby steps. Success is not built overnight; you have to work your way up the totem pole of fleecing people. Start small: persistently ask basic, RTFM-grade newbie questions while assigning yourself pompous, self-aggrandising titles like Asterisk Engineer. Keep it up, and you'll be crashing national economies with fraudulently constructed multi-billion dollar securitised debt tranches in no time. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requires
Alex you are my role model... Next time I'm in Atlanta - let's do lunch! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, July 18, 2011 9:08 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Requires On 07/18/2011 09:00 AM, Robert Huddleston wrote: Boy if only it was Enron :) Baby steps. Success is not built overnight; you have to work your way up the totem pole of fleecing people. Start small: persistently ask basic, RTFM-grade newbie questions while assigning yourself pompous, self-aggrandising titles like Asterisk Engineer. Keep it up, and you'll be crashing national economies with fraudulently constructed multi-billion dollar securitised debt tranches in no time. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
No. The only database stuff we do is MySQL CDRs Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Monday, July 18, 2011 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 Hi Eric, are you using ODBC? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 18 July 2011 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Monday, July 18, 2011 7:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Seg Faults with 1.6.2.19 Hi, is anyone else having problems with the reload command crashing Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have this problem but 1.6.2.19 after a few reloads is just dumping and restarting. We experienced the same thing. After a few reloads, Asterisk crashes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4] Minimal installation?
Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? = /bin/asterisk /etc/asterisk/ asterisk.conf logger.conf modules.conf sip.conf extensions.conf voicemail.conf /etc/init.d/asterisk /usr/lib/asterisk/modules/ /var/lib/asterisk/agi-bin/moh - /var/lib/asterisk/sounds/moh /var/lib/asterisk/sounds/ /var/lib/asterisk/agi-bin/static-http/ /var/spool/asterisk/ = Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX with SIP
So Steve I looked this, but, i didn't understood the difference between enable T.38 and T.38 Gateway, this site ttp://www.voip-info.org/wiki/view/T.38 talk --Asterisk *1.6* support G.711 and T.38 FAX origination and termination. T.38 gateway features are still in development. -- I know that Asterisk 1.10-beta1 already work with T.38 gateway, but the ask is, i need T.38 gateway to fax works? and how i know if T.38 is enable? I put on sip.conf and sip_general_custom.conf the following entry... t38pt_udptl=yes Is right? Thank you!! 2011/7/18 Steve Davies davies...@gmail.com On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org The magic sauce that you need is T.38 - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
On 07/18/2011 08:07 AM, Steve Davies wrote: On 18 July 2011 14:05, Lee Archerlee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. That should be the case, yes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, affordable x86 devices?
- Original Message - Hello I'd like to build a compact, affordable, fanless x86 solution to handle my home landline. I know about the following two platforms: 1. www.pcengines.ch/alix.htm alix1d + case 100€ Does Availability 500 mean that it's just not possible to buy just one item? 2. www.soekris.com/products.html?limit=all net4501-30 Board and Case $175.00 Is the net4501 powerful enough to run Asterisk, considering that I'll use an external VoIP gateway to connect it to my landline? Are there other manufacturers I should know about? You may be interested in the 'Blackbochs' SBC available here***: http://www.rockbochs.com/products/blackbochs-sbc --Tim ***Disclaimer: I work for the company offering this product. My intention is not to SPAM the list with adverts, etc, but rather to provide a useful answer within context of the OP. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Audio after attended tranfer
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an attended transfer. The transfer is going to an outbound number (normally AA on another IP PBX). the audio on the first transfer is fine. But if the user requests a transfer from AA to another department, I loose audio from Asterisk to the 2nd transfer. Based on the review of SIP packets, the second transfer issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not being handled properly by asterisk. I searched thru JIRA cases, but did not find anything like that. Any help would be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro issue under 1.8.5
- Original Message -
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I
missing
as
app_macro has been installed okay?
Macro was depreciated in 1.6 and most likely removed in 1.8.5
Removed, no. However in future version of Asterisk it will not
be
enabled in menuselect by default.
@OP: *CLI module load app_macro.so
Same problem even after performing the above load. module does
exist:
Watch the console carefully for errors when you run that command.
They
should tell you exactly what's wrong.
Also, it may help to inspect the differences in apps/app_macro.c
between
1.8.3 and 1.8.5.
Well it seems like its getting worse!
[Jul 18 11:36:00] WARNING[28936]: pbx.c:4071 pbx_extension_helper: No
application 'Playback' for extension (home, 400, 1)
Looking in pbx.c it would appear it cannot find the application in
some sort of cache:
if (!e-cached_app)
e-cached_app = pbx_findapp(e-app);
app = e-cached_app;
ast_unlock_contexts();
if (!app) {
ast_log(LOG_WARNING, No application '%s' for extension (%s, %s,
%d)\n, e-app, context, exten, priority);
return -1;
}
Any thoughts ?
Okay, I cleared out /usr/lib/asterisk/modules plus my build directory and
started with a fresh extract of asterisk tar file. This time all seems a lot
better apart from:
[Jul 18 12:25:38] ERROR[14082] pbx.c: Function CALLERID not registered
for which I need to add into modules.conf:
load = func_callerid.so
Why is this need now as it was not necessary in 1.8.3 ?
--
Thanks, Phil
--
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Re: [asterisk-users] Seg Faults with 1.6.2.19
Similar if not the same behavior still observed as of 1.8.5.0 with FreePBX. See https://issues.asterisk.org/jira/browse/ASTERISK-17498 -Vladimir On 7/18/2011 8:07 AM, Steve Davies wrote: On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seg Faults with 1.6.2.19
Hi Kevin, the ticket below was closed as it doesn't happen with 1.8. It can't be related to my ODBC connections if others are having it. Should a new ticket be opened? Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: 18 July 2011 15:10 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19 On 07/18/2011 08:07 AM, Steve Davies wrote: On 18 July 2011 14:05, Lee Archerlee.arc...@thebigword.com wrote: Seems to be an already reported problem but since no more fixes for 1.6 it's back to 1.6.2.18.2 for me. https://issues.asterisk.org/jira/browse/ASTERISK-18103 Regards Lee If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. That should be the case, yes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_gtalk load error
Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1 root root 281994 Jul 18 16:14 /usr/local/lib/libiksemel.a -rwxr-xr-x 1 root root822 Jul 18 16:14 /usr/local/lib/libiksemel.la lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so - libiksemel.so.3.1.1 lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so.3 - libiksemel.so.3.1.1 -rwxr-xr-x 1 root root 165132 Jul 18 16:14 /usr/local/lib/libiksemel.so.3.1.1 and checking whether they have been linked okay: ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff01523000) libiksemel.so.3 = /usr/local/lib/libiksemel.so.3 (0x2b6fbeb09000) libssl.so.6 = /lib64/libssl.so.6 (0x2b6fbed15000) libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b6fbef62000) libpthread.so.0 = /lib64/libpthread.so.0 (0x2b6fbf2b3000) libc.so.6 = /lib64/libc.so.6 (0x2b6fbf4ce000) libgnutls.so.13 = /usr/lib64/libgnutls.so.13 (0x2b6fbf827000) libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2b6fbfaab000) libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b6fbff6f000) libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b6fc0171000) libdl.so.2 = /lib64/libdl.so.2 (0x2b6fc0396000) libz.so.1 = /usr/lib64/libz.so.1 (0x2b6fc059b000) /lib64/ld-linux-x86-64.so.2 (0x003ac420) libgcrypt.so.11 = /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000) libgpg-error.so.0 = /usr/lib64/libgpg-error.so.0 (0x2b6fc0a21000) libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2b6fc0c25000) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b6fc0e2d000) libresolv.so.2 = /lib64/libresolv.so.2 (0x2b6fc102f000) libselinux.so.1 = /lib64/libselinux.so.1 (0x2b6fc1245000) libsepol.so.1 = /lib64/libsepol.so.1 (0x2b6fc145d000) Any thoughts on why this is happening as I could not find many references to it when searching ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_gtalk load error
- Original Message - From: --[ UxBoD ]-- ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 18, 2011 11:42:25 AM Subject: [asterisk-users] chan_gtalk load error Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1 root root 281994 Jul 18 16:14 /usr/local/lib/libiksemel.a -rwxr-xr-x 1 root root 822 Jul 18 16:14 /usr/local/lib/libiksemel.la lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so - libiksemel.so.3.1.1 lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so.3 - libiksemel.so.3.1.1 -rwxr-xr-x 1 root root 165132 Jul 18 16:14 /usr/local/lib/libiksemel.so.3.1.1 and checking whether they have been linked okay: ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff01523000) libiksemel.so.3 = /usr/local/lib/libiksemel.so.3 (0x2b6fbeb09000) libssl.so.6 = /lib64/libssl.so.6 (0x2b6fbed15000) libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b6fbef62000) libpthread.so.0 = /lib64/libpthread.so.0 (0x2b6fbf2b3000) libc.so.6 = /lib64/libc.so.6 (0x2b6fbf4ce000) libgnutls.so.13 = /usr/lib64/libgnutls.so.13 (0x2b6fbf827000) libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2b6fbfaab000) libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b6fbff6f000) libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b6fc0171000) libdl.so.2 = /lib64/libdl.so.2 (0x2b6fc0396000) libz.so.1 = /usr/lib64/libz.so.1 (0x2b6fc059b000) /lib64/ld-linux-x86-64.so.2 (0x003ac420) libgcrypt.so.11 = /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000) libgpg-error.so.0 = /usr/lib64/libgpg-error.so.0 (0x2b6fc0a21000) libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2b6fc0c25000) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b6fc0e2d000) libresolv.so.2 = /lib64/libresolv.so.2 (0x2b6fc102f000) libselinux.so.1 = /lib64/libselinux.so.1 (0x2b6fc1245000) libsepol.so.1 = /lib64/libsepol.so.1 (0x2b6fc145d000) Any thoughts on why this is happening as I could not find many references to it when searching ? -- Thanks, Phil Do you have res_jabber installed? -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org The_Boy_Wonder in #asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_gtalk load error
- Original Message - - Original Message - From: --[ UxBoD ]-- ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 18, 2011 11:42:25 AM Subject: [asterisk-users] chan_gtalk load error Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1 root root 281994 Jul 18 16:14 /usr/local/lib/libiksemel.a -rwxr-xr-x 1 root root 822 Jul 18 16:14 /usr/local/lib/libiksemel.la lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so - libiksemel.so.3.1.1 lrwxrwxrwx 1 root root 19 Jul 18 16:14 /usr/local/lib/libiksemel.so.3 - libiksemel.so.3.1.1 -rwxr-xr-x 1 root root 165132 Jul 18 16:14 /usr/local/lib/libiksemel.so.3.1.1 and checking whether they have been linked okay: ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff01523000) libiksemel.so.3 = /usr/local/lib/libiksemel.so.3 (0x2b6fbeb09000) libssl.so.6 = /lib64/libssl.so.6 (0x2b6fbed15000) libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2b6fbef62000) libpthread.so.0 = /lib64/libpthread.so.0 (0x2b6fbf2b3000) libc.so.6 = /lib64/libc.so.6 (0x2b6fbf4ce000) libgnutls.so.13 = /usr/lib64/libgnutls.so.13 (0x2b6fbf827000) libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2b6fbfaab000) libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2b6fbfcd9000) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2b6fbff6f000) libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2b6fc0171000) libdl.so.2 = /lib64/libdl.so.2 (0x2b6fc0396000) libz.so.1 = /usr/lib64/libz.so.1 (0x2b6fc059b000) /lib64/ld-linux-x86-64.so.2 (0x003ac420) libgcrypt.so.11 = /usr/lib64/libgcrypt.so.11 (0x2b6fc07af000) libgpg-error.so.0 = /usr/lib64/libgpg-error.so.0 (0x2b6fc0a21000) libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2b6fc0c25000) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2b6fc0e2d000) libresolv.so.2 = /lib64/libresolv.so.2 (0x2b6fc102f000) libselinux.so.1 = /lib64/libselinux.so.1 (0x2b6fc1245000) libsepol.so.1 = /lib64/libsepol.so.1 (0x2b6fc145d000) Any thoughts on why this is happening as I could not find many references to it when searching ? -- Thanks, Phil Do you have res_jabber installed? That would help :) Thanks David. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Minimal installation?
On Mon, Jul 18, 2011 at 03:20:03PM +0200, Gilles wrote: Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this, I'd like to know which directories/files are required for a basic install? Does this look right? = /bin/asterisk /usr/sbin , normally. But just the same. /etc/asterisk/ asterisk.conf logger.conf modules.conf sip.conf extensions.conf voicemail.conf Config files don't take that much space. Strip out comments and empty lines from the sample config files. Something along the lines of: sed -i -e 's/;.*//' -e '/^ *$/d' /etc/asterisk/*.conf /etc/init.d/asterisk /usr/lib/asterisk/modules/ Be sure to only include the ones you need. Finding which exactly may be tricky. /var/lib/asterisk/agi-bin/moh - /var/lib/asterisk/sounds/moh /var/lib/asterisk/sounds/ Only the ones you need . /var/lib/asterisk/agi-bin/static-http/ If you actually use the asterisk httpd . /var/spool/asterisk/ = -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libss7 variables
Hello! I am wondering if the Libss7 add-on for Asterisk also translates ss7 variables into the dialplans for routing, accounting, etc? Thanks, Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the use for the agent password if login via exten
Dears; If I need to login using as agent using the AddQueueMember(team,) then what to be the second paramter? How to be written? For example, if the agent id is 8000 then it will be: AddQueueMember(CustomerSupport,Agent/8000) or something else? Regards Bilal --- you have 2 options, add an agent to the queue or add a registered ip phone( or pstn line) to the queue. in first option, your operator must enter a password to identify as agent. but next option does not need password. On Mon, Jul 11, 2011 at 3:06 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten = 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the use for the agent password if login via exten
El 18/07/11 18:03, bilal ghayyad escribió: Dears; If I need to login using as agent using the AddQueueMember(team,) then what to be the second paramter? How to be written? For example, if the agent id is 8000 then it will be: AddQueueMember(CustomerSupport,Agent/8000) or something else? Regards Bilal Hi, You're right, the syntax is correct to add an Agent interface to a Queue. You can always check the inside CLI help: *CLI core show application AddQueueMember Cheers, --- Este mensaje y sus anexos son para uso exclusivo de sus destinatarios y puede contener informacion confidencial y/o privada protegida legalmente. Si usted no es el destinatario, se le notifica que cualquier distribucion o reproduccion de este mensaje, o de cualquiera de sus anexos, esta estrictamente prohibida. Si usted ha recibido este mensaje por error, por favor notifiquenos inmediatamente y elimine su texto original, incluidos los anexos y destruya cualquier reproduccion del mismo. Las opiniones expresadas en este mensaje son responsabilidad exclusiva de quien las emite y no necesariamente reflejan la posicion de Millenium Phone Center S.A, ni comprometen la responsabilidad institucional por el uso que el destinatario haga de las mismas. --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX with SIP
Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX with SIP
I resoundingly second that. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 18, 2011, at 11:12 PM, C F shma...@gmail.com wrote: Short answer is: dont use it. For the long answer wait for others to answer that. On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote: Hello guys I need some help to do works FAX using SIP, anybody know the secret to this? Have asterisk 1.6. Thanks!! -- Enviado do meu celular Eduardo Carpes E-mail: car...@bsd.com.br www.freebsd.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
On Mon, Feb 21, 2011 at 1:21 AM, Vladimir Mikhelson v...@mikhelson.comwrote: William, It still looks like something is not properly set with your account on Google Voice. Have you had a chance to follow the recommendations I gave you earlier in the thread? If the account is properly set the dial string will need to look like this, gtalk/jabber-conf-section-name/+$OUTNUM$@voice.google.com where $OUTNUM$ is a called number in the international format. On the receiving end the call will come with an empty CID Number, but with the CID Name which looks like this: +1551...@voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM= Just cut all prior to @ as a CID Number. See https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Also you do not need to wait 5 seconds. 1 or 2 is sufficient. -Vladimir This is a really old thread but I am having the same issues as William was having. The incoming call just doesn't hit the context in extensions.conf. I see the call come in on jabber...but I've tried almost 4-5 different variations of handling the call in extensions.conf from examples on the web, but nothing happens. I'm on 1.8.5.0. BTW, there is no Google Voice involved. and I'm calling from from a gmail based gtalk client. Also, I can successfully make an outbound call. Just the inbound isn't working :( Any help please? Currently my incoming dial-plan is: [gtalk-in] exten = s,1,Answer() same = n,Wait(2) same = n,SendDTMF(1) same = n,Dial(SIP/2000,20) and I have tried a whole bunch of stuff in jabber.conf and gtalk.conf but nothing seems to cut it. I have also tried using matching my email address (called gtalk a/c) to match in the exten as opposed to 's' extension and that doesn't work either. gtalk.conf -- [general] context=gtalk-in bindaddr=0.0.0.0 externip=my external address allowguest=yes [guest] disallow=all allow=ulaw context=gtalk-in connection=asterisk [aeg74] username=ae...@gmail.com disallow=all allow=ulaw context=gtalk-in connection=asterisk jabber.conf [general] debug=yes autoprune=yes autoregister=yes [asterisk] type=client serverhost=talk.google.com username=all.efor...@gmail.com/Talk secret=my secret port=5222 ; Port to use defaults to 5222 usetls=yes ; Use tls or not usesasl=yes ; Use sasl or not buddy=ae...@gmail.com status=available statusmessage=On Asterisk timeout=100 *This is the debug on jabber* JABBER: asterisk INCOMING: iq type=set to= all.efor...@gmail.com/Talk17BFE21F id=CA051C15DD949454 from= ae...@gmail.com/gmail.320B5151jin:jingle action=session-initiate sid=c1901211999 initiator=ae...@gmail.com/gmail.320B5151 xmlns:jin=urn:xmpp:jingle:1jin:content name=audio creator=initiatorrtp:description media=audio xmlns:rtp=urn:xmpp:jingle:apps:rtp:1rtp:payload-type id=103 name=ISAC clockrate=16000rtp:parameter name=bitrate value=32000//rtp:payload-typertp:payload-type id=104 name=ISAC clockrate=32000rtp:parameter name=bitrate value=56000//rtp:payload-typertp:payload-type id=119 name=ISACLC clockrate=16000rtp:parameter name=bitrate value=4//rtp:payload-typertp:payload-type id=99 name=speex clockrate=16000rtp:parameter name=bitrate value=22000//rtp:payload-typertp:payload-type id=97 name=IPCMWB clockrate=16000rtp:parameter name=bitrate value=8//rtp:payload-typertp:payload-type id=9 name=G722 [Jul 18 23:36:15] JABBER: asterisk INCOMING: clockrate=16000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=102 name=iLBC clockrate=8000rtp:parameter name=bitrate value=13300//rtp:payload-typertp:payload-type id=98 name=speex clockrate=8000rtp:parameter name=bitrate value=11000//rtp:payload-typertp:payload-type id=3 name=GSM clockrate=8000rtp:parameter name=bitrate value=13200//rtp:payload-typertp:payload-type id=100 name=EG711U clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=101 name=EG711A clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=0 name=PCMU clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=8 name=PCMA clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=117 name=red clockrate=8000/rtp:payload-type id=106 name= [Jul 18 23:36:15] JABBER: asterisk INCOMING: telephone-event clockrate=8000//rtp:descriptionp:transport xmlns:p= http://www.google.com/transport/p2p//jin:content/jin:jingleses:session type=initiate id=c1901211999 initiator=ae...@gmail.com/gmail.320B5151 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho= http://www.google.com/session/phone;pho:payload-type id=103 name=ISAC bitrate=32000 clockrate=16000/pho:payload-type id=104 name=ISAC bitrate=56000 clockrate=32000/pho:payload-type id=119 name=ISACLC bitrate=4 clockrate=16000/pho:payload-type id=99 name=speex bitrate=22000 clockrate=16000/pho:payload-type id=97
