Re: [asterisk-users] Bridging an Answered call in Asterisk with another call
Jayesh, Personally I haven't worked on Congbridge :).
Confbridge has evolved a lot in 10.X. So probably you should have no issues
using it.
On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar jayesh.v...@gmail.comwrote:
Thank you Satish. I was also thinking on similar lines. I was just
wondering if there was any mechanism with which we can bridge a new call
with the existing running call if the Call-ID of the call is known !!
I can definitely use the confbridge application for the same right; as I
am working on Asterisk10. What do you suggest??
Thanks again,
--- Jayesh
On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot
satish4aster...@gmail.comwrote:
Make your user wait in a *Meetme* and then call your destination number
through AMI and once he answers, place him in the same *Meetme*.
e.g. Assuming your destination is SIP extension, have something like...
Action: Originate
Channel: SIP/{your_destination_here}
Application: MeetMe
Data: {your_meetme_number_here}
Hope this helps.
--Satish Barot
On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar jayesh.v...@gmail.comwrote:
Hello All,
I need to know a way of connecting an Answered call in Asterisk to
another call which was triggered by an AMI. I have a scenario as follows:
1) User dials 123 from a touch screen Polycom phone.
2) Call comes to Asterisk and Asterisk answers the call and asks for PIN
number.
3) Once the PIN is validated, Asterisk sends a User Event through AMI
which invokes a browser in the Polycom phone.
4) The Browser will have a Text-Box to Enter the destination number
where the caller wants to be bridged.
5) The caller enters this number in the browser which is sent as a
Originate command to Asterisk through the AMI. Please note Asterisk does
not get this number as DTMF events !!
6) Now, I need to BRIDGE this originated call from the AMI with the
actual caller who is already present in Answered state in Asterisk probably
listening to some music.
Is there any straightforward application or function to achieve this in
Asterisk.
Any ideas or directions will be of great help !!
Thanks,
--- Jayesh
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[asterisk-users] Sip insecure
Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza -- _ -- 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] Sip insecure
2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Sip insecure
They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.com wrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Freepbx and Http proxies
Hi, Tough Freepbx is not the main focus of this list, may I ask if Freepbx and its End Point Manager module can work in an environment with an HTTP proxy ? In my testing, everything works OK but one thing: I can't upload End Point product list : in End Point Configuration tab, when I click over Check for Updates button, I get this: Not able to connect to repository. Using local master file instead. Aborting Brand Downloads. Can't Get Master File, Assuming Timeout Issues! Learn how to manually upload packages here (it's easy!): Click Here! Any pointer would be greatly appreciated. Regards -- _ -- 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] Sip insecure
My main box is asterisk 1.8 and there are two boxes, one asterisk 1.8 and other 1.4 with 1.4, I don't need to define fromuser=username but in 1.8 I can't make call without it the problem in defining fromuser is, it overrides the callerid Main box has same settings for both peers Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:26 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] ConfBridge and SLA
MeetMe seems to be deprecated as of 10.2.1 and replaced by ConfBridge. Does anyone know if SLA (which is built on MeetMe) will use/be ported to ConfBridge? Thanks. -- _ -- 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] All circuits are busy now on outgoing trunk call
Hi, Thanks for the support. Issue solved. Somehow the routes on the fxo gw were not working. On 3/21/12, James Mutuku listmut...@gmail.com wrote: Hi, I have configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see. On 3/21/12, SamyGo govoi...@gmail.com wrote: 404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku listmut...@gmail.com wrote: I am setting up asterisk-fxo gw. 404 Not Found (User not found) means the user is not found, but I don't need to have extensions or authentication on the fxo gw On 3/21/12, Michael L. Young myo...@acsacc.com wrote: [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251 ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 I think the 404 Not Found being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- 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 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- 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 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- 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] Sip insecure
I've figured this out using match_auth_username =yes Thanks Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:33 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: My main box is asterisk 1.8 and there are two boxes, one asterisk 1.8 and other 1.4 with 1.4, I don't need to define fromuser=username but in 1.8 I can't make call without it the problem in defining fromuser is, it overrides the callerid Main box has same settings for both peers Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:26 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: They don't require authentication of invites which I do need Regards, Zohair Raza On Thu, Mar 22, 2012 at 3:20 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/3/22 Zohair Raza engineerzuhairr...@gmail.com Hi, How to allow registered sip users to call without re-authentication insecure =yes/very are deprecated in 1.8 I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch. Please help Regards, Zohair Raza There are other options, like invite and port to be used when you trust the IP of the caller. Leandro -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
Hi gang, I've put 10.X on about 15 different VM's now, but I've run into a buzzsaw on this one and my google-fu has failed me Output of make CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/usr/local/src/asterisk-10.1.3/menuselect' make[1]: `menuselect' is up to date. make[1]: Leaving directory `/usr/local/src/asterisk-10.1.3/menuselect' [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized collect2: ld returned 1 exit status make[1]: *** [astdb2sqlite3] Error 1 make: *** [utils] Error 2 uname -a Linux 2.6.16.60-0.21-smp #1 SMP Tue May 6 12:41:02 UTC 2008 x86_64 x86_64 x86_64 GNU/Linux Welcome to SUSE Linux Enterprise Server 10 SP2 (x86_64) - Kernel \r (\l). Any suggestions? Thanks in advance Danny Nicholas -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
Danny Nicholas wrote: libstdc++.so: file not recognized: File format not recognized Google shows: Very likely you try to link a 64-bit executable with a 32-bit library or vice versa. http://www.groupsrv.com/linux/about144074.html 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] Problem installing asterisk 10.1.3 on SUSE
So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 22, 2012 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE Danny Nicholas wrote: libstdc++.so: file not recognized: File format not recognized Google shows: Very likely you try to link a 64-bit executable with a 32-bit library or vice versa. http://www.groupsrv.com/linux/about144074.html 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 -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
I am running Asterisk 10.X on OpenSuse 12.x in virtual machines and it is working good. I had issues prior to the 12.x version when trying to VM Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Danny Nicholas da...@debsinc.com Sent: Thursday, March 22, 2012 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, March 22, 2012 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE Danny Nicholas wrote: libstdc++.so: file not recognized: File format not recognized Google shows: Very likely you try to link a 64-bit executable with a 32-bit library or vice versa. http://www.groupsrv.com/linux/about144074.html 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 -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
On 22-03-12 16:47, Danny Nicholas wrote: So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3! Depends on how you compile it or what version you downloaded. Both are possible. Asterisk 10 32bit requires a 32bit OS and Asterisk 10 64bit requires a 64bit OS. I didn't understand pox on sqlite3!. Please elaborate. Regards, Patrick -- _ -- 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] Asterisk generating backtrace
On 03/21/2012 08:04 PM, Jonas Kellens wrote: Hello, when generating backtrace I get following output : /[root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt asterisk: No such file or directory./ /warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff00799000/ What am I doing wrong ? Hello list, please help. I want to know what went wrong with Asterisk, so I want to generate a backtrace. If asterisk is not the correct value for the -se flag, what could it be ? Which parameter is expected here ? Thanks, Jonas. -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
On Thursday 22 March 2012, Danny Nicholas wrote: Hi gang, I've put 10.X on about 15 different VM's now, but I've run into a buzzsaw on this one and my google-fu has failed me Output of make CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/usr/local/src/asterisk-10.1.3/menuselect' make[1]: `menuselect' is up to date. make[1]: Leaving directory `/usr/local/src/asterisk-10.1.3/menuselect' [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized collect2: ld returned 1 exit status make[1]: *** [astdb2sqlite3] Error 1 make: *** [utils] Error 2 You're most probably missing a library or -devel package on the 64-bit side, but you have the 32-bit version kicking around and this is confusing things. -- AJS Answers come *after* questions. -- _ -- 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] Asterisk generating backtrace
On Wed, Mar 21, 2012 at 3:04 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, when generating backtrace I get following output : [root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt asterisk: No such file or directory. warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff00799000 What am I doing wrong ? Kind regards, Jonas. maybe you need to quote the core file name with those chars. Try renaming the core file to something simple -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Asterisk generating backtrace
Have you read the backtrace.txt included in the doc/ directory Asterisk source code? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, March 22, 2012 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk generating backtrace On 03/21/2012 08:04 PM, Jonas Kellens wrote: Hello, when generating backtrace I get following output : [root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt asterisk: No such file or directory. warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff00799000 What am I doing wrong ? Hello list, please help. I want to know what went wrong with Asterisk, so I want to generate a backtrace. If asterisk is not the correct value for the -se flag, what could it be ? Which parameter is expected here ? Thanks, Jonas. -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
Confusion? I'm looking at the Asterisk Release download directory and only see asterisk-10.1.3.tar.gz, not asterisk-10.1.3-32.tar.gz and asterisk-10.1.3.64.tar.gz. is this a configuration or make option? As for the pox on sqlite3, I have had varying degrees of misery with this - on one install I had to resort to deleting the astdb table using bash in safe_asterisk because asterisk crashed every time it tried to do create if not exists table astdb -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists Sent: Thursday, March 22, 2012 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE On 22-03-12 16:47, Danny Nicholas wrote: So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3! Depends on how you compile it or what version you downloaded. Both are possible. Asterisk 10 32bit requires a 32bit OS and Asterisk 10 64bit requires a 64bit OS. I didn't understand pox on sqlite3!. Please elaborate. Regards, Patrick -- _ -- 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] Asterisk generating backtrace
I'm following https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-PreparingAsteriskToProduceCoreFilesOnCrash But there is nowhere information on possible error-messages that you can get... Do you know what is wrong when gdb says asterisk: No such file or directory ? I would like to here from you. Jonas. On 03/22/2012 05:12 PM, Eric Wieling wrote: Have you read the backtrace.txt included in the doc/ directory Asterisk source code? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, March 22, 2012 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk generating backtrace On 03/21/2012 08:04 PM, Jonas Kellens wrote: Hello, when generating backtrace I get following output : [root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt asterisk: No such file or directory. warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7fff00799000 What am I doing wrong ? Hello list, please help. I want to know what went wrong with Asterisk, so I want to generate a backtrace. If asterisk is not the correct value for the -se flag, what could it be ? Which parameter is expected here ? Thanks, Jonas. -- _ -- 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] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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] Problem installing asterisk 10.1.3 on SUSE
On 22-03-12 17:26, Danny Nicholas wrote: Confusion? I'm looking at the Asterisk Release download directory and only see asterisk-10.1.3.tar.gz, not asterisk-10.1.3-32.tar.gz and asterisk-10.1.3.64.tar.gz. is this a configuration or make option? As for the pox on sqlite3, I have had varying degrees of misery with this - on one install I had to resort to deleting the astdb table using bash in safe_asterisk because asterisk crashed every time it tried to do create if not exists table astdb Heh confusion indeed. I meant that the sources are architecture independent (assuming they can be compiled on e.g. 32bit and 64bit platforms) and that the resulting binaries are obviously different and tied to the platform they are built for. I haven't seen your compilation error before so can be of little help. Perhaps check if that library is not corrupt (reinstall to make sure). If nothing else, why not rm -rf the asterisk source tree and start from scratch. Sometimes things get in an odd state and whenever I see something unexplainable that that's what I usually do. If compilation keeps failing in different places without any apparent reason then maybe it's time to get out the rescue cd and run memtest86+, check hardware etc. Regards, Patrick -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
On Thursday 22 March 2012, Danny Nicholas wrote: Confusion? I'm looking at the Asterisk Release download directory and only see asterisk-10.1.3.tar.gz, not asterisk-10.1.3-32.tar.gz and asterisk-10.1.3.64.tar.gz. is this a configuration or make option? Both 32-bit and 64-bit versions (and, indeed, the versions for non-Intel architectures) are built from exactly the same Source Code. Which version gets built is determined in distro-dependent ways: for instance, Gentoo has use flags which determine various things about how packages are built. I'm not familiar with SuSE, so you'll have to consult your documentation. However, the selection of what target you're building for most probably is determined either by exporting environment variables, or by passing parameters to the configure script. -- AJS Answers come *after* questions. -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
Just to update everyone, I did a make -i to ignore the error on the required make of astdb2sqlite3, but haven't done the make install because I can't bring the machine down to try it at this time. I have a hunch that either Asterisk will crash or I will have to do some sqlite voodoo to get it up and running. I know that the ASTDB functionality is important to lots of folks and functions, but you should be able to compile and start without it if you have a simple dialplan that doesn't use it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Thursday, March 22, 2012 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE On Thursday 22 March 2012, Danny Nicholas wrote: Confusion? I'm looking at the Asterisk Release download directory and only see asterisk-10.1.3.tar.gz, not asterisk-10.1.3-32.tar.gz and asterisk-10.1.3.64.tar.gz. is this a configuration or make option? Both 32-bit and 64-bit versions (and, indeed, the versions for non-Intel architectures) are built from exactly the same Source Code. Which version gets built is determined in distro-dependent ways: for instance, Gentoo has use flags which determine various things about how packages are built. I'm not familiar with SuSE, so you'll have to consult your documentation. However, the selection of what target you're building for most probably is determined either by exporting environment variables, or by passing parameters to the configure script. -- AJS Answers come *after* questions. -- _ -- 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] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
Solve it. :) Found this link: http://www.voip-info.org/wiki/view/SCCP-HOWTO2 Cheers, Alex 2012/3/22 Alexandre Rodrigues alex...@gmail.com: Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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] Freepbx and Http proxies
I've tried this in the past and while FreePBX and its base modules work fine in an http proxy environment, some applications like fop2 fail to connect properly as they obviously rely on direct connections via ajax using the browser as a client. That said, I've never tested the end point manager in this capacity. The error seems to indicate the module doesn't have connectivity when trying to access something on the internet or it's getting a 404 or something. Are you able to update and install freepbx modules in a similar manner from the same freepbx installation while it's running behind the proxy? That should tell you whether or not your problem is stemming from general connectivity issues and not just the end point module in that regard. Always helps to rule stuff out. John Knight Classic City Telco LLC Email: j...@classiccitytelco.com | Main: (706) 995-0200 Direct: (706) 995-0201 | Mobile: (706) 255-9203 CCT Enterprise Linux 6 is released! Click here to learn more. On 3/22/2012 7:31 AM, Olivier wrote: Hi, Tough Freepbx is not the main focus of this list, may I ask if Freepbx and its End Point Manager module can work in an environment with an HTTP proxy ? In my testing, everything works OK but one thing: I can't upload End Point product list : in End Point Configuration tab, when I click over Check for Updates button, I get this: Not able to connect to repository. Using local master file instead. Aborting Brand Downloads. Can't Get Master File, Assuming Timeout Issues! Learn how to manually upload packages here (it's easy!): Click Here! Any pointer would be greatly appreciated. Regards -- _ -- 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] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5
On Fri, Mar 16, 2012 at 11:13:14PM +0400, Vahan Yerkanian wrote: I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with Running dahdi_cfg: DAHDI startup failed: Input/output error Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo? Vahan, Just closing out this public thread. dahdi-linux 2.6.1 will contain what fixed this issue on your machine. I committed onto both trunk [1] and onto the 2.6 branch [2]. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10559 [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10565 Thanks again for your help, SHaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Asterisk generating backtrace
On Thu, Mar 22, 2012 at 6:29 PM, Jonas Kellens jonas.kell...@telenet.be wrote: I'm following https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-PreparingAsteriskToProduceCoreFilesOnCrash But there is nowhere information on possible error-messages that you can get... Do you know what is wrong when gdb says asterisk: No such file or directory ? I would like to here from you. Jonas. not a gdb expert but I suspect that you should provide the full path to the asterisk executable (like -se `which asterisk`). HTH, Ioan -- _ -- 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] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
i can make bouns mints to asterisk and elastix just give me the ips and i will add mints send the ip or host to civic_t...@yahoo.com From: Alexandre Rodrigues alex...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 22, 2012 6:28 PM Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 to Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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 --
Re: [asterisk-users] SendText causes Retransmission errors
On 03/20/2012 01:08 PM, Matt Hamilton wrote: Date: Mon, 19 Mar 2012 10:31:52 -0500 From: kpflem...@digium.com 502 10.0.1.103 10.0.1.57 Request: CANCEL sip:104@10.0.1.57:5060 Why did Asterisk CANCEL the call here? I assume it's part of the SLA implementation. As I mentioned in my original email, I'm using SendText to send a text message when the user picks up a line in a SLA setup. In this case, ext 124 is calling 104, and one of the lines on 104 is picking it up. Asterisk is connecting to that line and cancelling the first request?? (just guessing) same = n,SendText(hi) same = n,SLAStation(4*104_line104) *503 (for 493) 10.0.1.57 10.0.1.103 Status: 200 OK* 524 (503) 10.0.1.57 10.0.1.103 Request: ACK sip:8*104_line104@10.0.1.103:5060 This appears to be broken. The listing here claims this ACK is in response to the '200 OK' in packet 503, which itself was a final response to the MESSAGE request in packet 493. However, MESSAGE requests do not use ACK for a three-way handshake like INVITE requests do. In addition, this packet is going the wrong direction to be an ACK for packet 503, since it's going the same direction as packet 503 did. I use Wireshark to capture the packets, and Wireshark is reporting it that way; i.e. saying that Request Frame for the ACK is the OK (for MESSAGE). I guess it's incorrect. The order and direction of messages I posted in my previous email are taken directly from Wireshark. Frame 15 is MESSAGE Frame 19 is OK (for MESSAGE) Frame 20 is ACK (Wireshark is saying the Request Frame is 20 ??) I tried to post the full SIP capture here, but it got rejected because of the size of the post (about 280k). Yep, that's a lot. The next step is probably to open an issue in our issue tracker and upload the capture file there (feel free to compress it first to save time and space). -- 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] Problem installing asterisk 10.1.3 on SUSE
On 03/22/2012 12:49 PM, Danny Nicholas wrote: Just to update everyone, I did a make -i to ignore the error on the required make of astdb2sqlite3, but haven't done the make install because I can't bring the machine down to try it at this time. I have a hunch that either Asterisk will crash or I will have to do some sqlite voodoo to get it up and running. I know that the ASTDB functionality is important to lots of folks and functions, but you should be able to compile and start without it if you have a simple dialplan that doesn't use it. Parts of Asterisk itself use the Asterisk DB, that's the primary reason it exists. If you don't have a working Asterisk DB, then you are not likely to have a working Asterisk system. -- 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
[asterisk-users] AgiOperations.getData--- all digits doubled
Hello, When I input 4165764323, the AgiOperations.getData receives this number as 44116655776644332233. This is happening for almost every other call. Do you have any idea what's wrong maybe? Thanks for help -- Song -- _ -- 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] Official numbering plan - where to get?
I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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] Official numbering plan - where to get?
Although I do feel that 100+ Euros/month is more than most of us could manage, I don't think a one-time list is of much value. I would be interested in establishing a database if there was interest from enough users for a modest subscription price. --Don Don Kelly PCF Corp People Come First 651 842-1000 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Thursday, March 22, 2012 5:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Official numbering plan - where to get? I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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] Problem installing asterisk 10.1.3 on SUSE
On Thursday 22 Mar 2012, Danny Nicholas wrote: Hi gang, I've put 10.X on about 15 different VM's now, but I've run into a buzzsaw on this one and my google-fu has failed me Output of make CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/usr/local/src/asterisk-10.1.3/menuselect' make[1]: `menuselect' is up to date. make[1]: Leaving directory `/usr/local/src/asterisk-10.1.3/menuselect' [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized Try: file astdb2sqlite3.o file db1-ast/*.o file /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so (with the appropriate paths for the first two) and see if they're the same (32 or 64-bit) architecture. The third is likely to be 64-bit anyway, what are the first two? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Official numbering plan - where to get?
Is it a problem to parse rates from said 10 providers and create database with all their info? Anyways, speaking of this as a service... I have at least 2 clients, who would love such service: some kind of daily (maybe more often) updated database, which automatically normalizes rates and provides output in parseable format. Maybe even that could include some interactive page, for providers which offer cheaper rates for higher call volumes. But of course 100 Euros/month will be too much for such service. AND some kind of integration with Starbilling will make the whole world happy. BR Don Kelly писал 23.03.2012 01:00: Although I do feel that 100+ Euros/month is more than most of us could manage, I don't think a one-time list is of much value. I would be interested in establishing a database if there was interest from enough users for a modest subscription price. --Don Don Kelly PCF Corp People Come First 651 842-1000 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Thursday, March 22, 2012 5:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Official numbering plan - where to get? I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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
