Re: [asterisk-users] Asterisk unresponsive
On 04/18/2011 06:36 PM, Paul Belanger wrote: On 11-04-18 09:46 AM, Jonas Kellens wrote: Asterisk freezed and only a reboot of the whole server fixed this. Any command on the Asterisk CLI was not executed because Asterisk was too busy processing all of these messages that you see in the debug log. What is the origin of these messages ? Sounds like a deadlock[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock Hello, the core dump file is saved in the /tmp directory. Of course this morning the file was automatically deleted. A deadlock, what can be causing this ? In the past I've posted another thread on this list about such a deadlock and the advice then was to upgrade. I've upgraded from 1.6.2.10 to 1.6.2.16.1, but all of a sudden I have a deadlock again. I've checked my verbose and debug log files, but I see nothing special. There were some conversations going on... nothing abnormal. Kind regards, 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
[asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND
Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe. Kind Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. 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
[asterisk-users] ConfBridge and AGI
Hello List, Is it possible to run an AGI script in backgroung for all the associated SIP channels in ConfBridge Application? If yes how? This can be done using 'b' parameter in MeetMe for non SIP channels. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. 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] R: No Internet, no asterisk
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- 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] No voice in MeetMe for SIP with AGI_BACKGROUND
In article 2658e54b540d284981ea57e6a549ea70abd1f7c...@inblrk77m1msx.in002.siemens.net, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe. The fact that background AGI in meetme only works with Zap channels is a consequence of the original design of Meetme. See these two old posts: http://lists.digium.com/pipermail/asterisk-users/2004-August/050725.html http://lists.digium.com/pipermail/asterisk-users/2004-August/050776.html You will need to change to a different approach to solve your requirement. Could you explain your original requirement? Then people on this list may be able to suggest an alternative way to do it. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] R: No Internet, no asterisk
On Tuesday 19 Apr 2011, Niccolò Belli wrote: A caching nameserver is not a viable solution because I want it working even after a month without internet access. Then just make your local nameserver authoritative for the domain in question. You can always firewall off port 53, if the nameserver faces outward and you're scared of breaking the Internet for other people :) P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? The most likely explanation is, the developers either have more reliable Internet connections (which maybe they need for SIP or IAX trunks) or have configured their nameservers properly. Remember, even a short how I fixed it for myself post counts as doing something useful for the community. -- 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] R: No Internet, no asterisk
I have enabled DNS manager in /etc/asterisk/dnsmgr.conf. It helps me. On 19.04.2011 14:05, Niccolò Belli wrote: Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- 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] R: R: No Internet, no asterisk
srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. Make sure you don't have ANY reference to domain names in your sip.conf, only IPs, and eventually try to specify the port as described above. I didn't tried this myself but I think this should be the way to do it (srvlookup=no). Regards, ALEX -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Niccolò Belli Inviato: martedì 19 aprile 2011 12:05 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: No Internet, no asterisk Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- 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] ConfBridge and AGI
Hi Rajib, this is your second post on Meetme with SIP channel and AGI script, Can you provide your requirement to run an AGI for Meetme , what you want to run an AGI with meetme. in confbridge there is nothing option for running AGI in background mode. let us know what you want to do exactly, on that basis people of group can help you. regards Dhaval On Tue, Apr 19, 2011 at 2:13 PM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, Is it possible to run an AGI script in backgroung for all the associated SIP channels in ConfBridge Application? If yes how? This can be done using ‘b’ parameter in MeetMe for non SIP channels. Regards, Rajib *Rajib Deka* SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com -- Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. 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 -- _ -- 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] Good by asterisk 1.4? Please not.
@digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) Kristijan 2011/4/15 Julian Lyndon-Smith aster...@dotr.com: 1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;) https://issues.asterisk.org/view.php?id=18951 Julian On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote: You know we don't have choise. I had remembered when we shifted 1.2 to first release of 1.4 and we had many issue. Same thing right now I'm dealing with 1.8 things take time to stabilized. Good luck!! -- Sent from my iPhone On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] R: No Internet, no asterisk
2011/4/19 Niccolò Belli darkbas...@gmail.com: Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Wouldn't a caching nameserver just return NXDOMAIN if it couldn't contact the authoritative server for that domain? -M -- _ -- 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] R: No Internet, no asterisk
Make sure ALL IP addresses of the system are in /etc/hosts, as well as the IP of your provider. Asterisk gets upset if it can't do a reverse lookup of an IP address on the system. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niccolò Belli Sent: Tuesday, April 19, 2011 6:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] R: No Internet, no asterisk Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- 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_dahdi under bristuff 0.4.0-RC11 not building under Linux 10.04.2 LTS
Hi all, I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server machine with 2.6.32-24-generic-pae kernel. The prereq.sh script executes without complaints (BTW on my system, libncurses-dev evaluates to libncurses5-dev and libz-dev evaluates to zlib1g-dev). With the asterisk 1.4.41 package that is installed, a make menuselect operation indicates that all dependencies are met as the chan_dahdi entry allows me to insert a * instead of the XXX for unmet dependencies. However, a compilation attempt fails and results in the following error: [CC] chan_dahdi.c - chan_dahdi.o chan_dahdi.c: In function ‘dahdi_hangup’: chan_dahdi.c:3169: warning: unused variable ‘outgoing’ chan_dahdi.c: In function ‘ss_thread’: chan_dahdi.c:6463: warning: unused variable ‘network’ chan_dahdi.c: At top level: chan_dahdi.c:9481: error: expected declaration specifiers or ‘...’ before ‘q931_call’ chan_dahdi.c: In function ‘pri_find_tei’: chan_dahdi.c:9484: error: dereferencing pointer to incomplete type chan_dahdi.c:9485: error: dereferencing pointer to incomplete type chan_dahdi.c:9486: error: dereferencing pointer to incomplete type chan_dahdi.c:9486: error: dereferencing pointer to incomplete type chan_dahdi.c:9486: error: ‘c’ undeclared (first use in this function) chan_dahdi.c:9486: error: (Each undeclared identifier is reported only once chan_dahdi.c:9486: error: for each function it appears in.) chan_dahdi.c: In function ‘pri_get_callonhold’: chan_dahdi.c:9494: error: dereferencing pointer to incomplete type chan_dahdi.c:9498: error: dereferencing pointer to incomplete type chan_dahdi.c:9498: error: dereferencing pointer to incomplete type chan_dahdi.c:9502: error: dereferencing pointer to incomplete type chan_dahdi.c: In function ‘pri_destroy_callonhold’: chan_dahdi.c:9508: error: dereferencing pointer to incomplete type chan_dahdi.c:9514: error: dereferencing pointer to incomplete type chan_dahdi.c:9514: error: dereferencing pointer to incomplete type chan_dahdi.c:9517: error: dereferencing pointer to incomplete type chan_dahdi.c:9517: error: dereferencing pointer to incomplete type chan_dahdi.c:9518: error: dereferencing pointer to incomplete type chan_dahdi.c:9523: error: dereferencing pointer to incomplete type chan_dahdi.c: In function ‘dahdi_sendtext’: chan_dahdi.c:14491: warning: implicit declaration of function ‘dahdi_tdd_sendtext’ chan_dahdi.c: At top level: chan_dahdi.c:14521: error: static declaration of ‘dahdi_tdd_sendtext’ follows non-static declaration chan_dahdi.c:14491: note: previous implicit declaration of ‘dahdi_tdd_sendtext’ was here make[1]: *** [chan_dahdi.o] Error 1 make: *** [channels] Error 2 Running a make clean, ./configure and running the make again does not help either. Any advice? Mos -- _ -- 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] Good by asterisk 1.4? Please not.
On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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] Good by asterisk 1.4? Please not.
Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ? If not, then I am stuck on my current version of 1.4, and will not be able to upgrade to either of those two versions, even for security fixes. Julian On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com wrote: On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] Good by asterisk 1.4? Please not.
Its not really had to install 1.6 or 1.8 on a test box, and see if a phone connects to it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, April 19, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good by asterisk 1.4? Please not. Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ? If not, then I am stuck on my current version of 1.4, and will not be able to upgrade to either of those two versions, even for security fixes. Julian On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com wrote: On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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_mobile: Dropping incompatible voice frame
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have no audio on chan_mobile but this message repeats continuously: Dropping incompatible voice frame on Mobile/DNA-54f4 of format slin since our native format has changed to 0x0 (nothing) Can somebody point me to the right direction? Asterisk SVN-branch-1.6.2-r313579 - -Stefan - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk2tr3YACgkQbQKZlCdPOMOLkwCfYm/jdPx3uOYdcvZ5XsZeKWAg sD8AoL4ygna6jWsKLY9sEwzU2VjRek/T =hbxJ -END PGP SIGNATURE- -- _ -- 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] RTP and Signalling Dropping
Hi I have a weird issue with a new 1.6.2.17.2 box. At random intervals it just stops responding to RTP and signalling (both SIP and IAX observed). All calls in progress lose audio both ways although the console shows the call legs still in progress. No signalling can be sent or is received. It is as though the server drops of the net for those protocols. I can still navigate the console. Killing an restarting Asterisk is the only way to bring it back. I can see nothing in the logs to indicate what is happening. The server is dual homed network one interface on a public address and the other interface on a private subnet that the phones sit on. It can do 100's even 1000's of calls before the issue happens and then BAM it drops off. The box is handling between 1500 - 3000 calls a day, mostly SIP and IAX with a small percentage of DADHI. Anyone any ideas what is going on or where to look next? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- 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] Good by asterisk 1.4? Please not.
The point I was trying to make was that *anyone* on 1.4 who uses ciscos will be forced to move to 1.6 or 1.8 if they want any security fixes applying , as the patches will go into 1.4 svn where the bug is present. IOW if you uses cisco's and 1.4 then that's the end of the line for you. No more patches *or* security fixes. People should be aware of that. But thanks for the helpful insight. Julian On 19 April 2011 16:18, William Stillwell will...@stillwellsoft.com wrote: Its not really had to install 1.6 or 1.8 on a test box, and see if a phone connects to it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, April 19, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good by asterisk 1.4? Please not. Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ? If not, then I am stuck on my current version of 1.4, and will not be able to upgrade to either of those two versions, even for security fixes. Julian On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com wrote: On 11-04-19 09:28 AM, Kristijan Vrban wrote: @digium 1. What happened with the 1.4 patches that still wait on issues.asterisk.org? e.g. issue #19108 Once a branch moves into security mode; no more bug fixes will be applied. If a security issue affects the 1.4 branch, a new release will be created containing only that fix. 2. What happened with bugfix patches for 1.4 made by the community. Will those be ignored now? (e.g. i have one more a memleak fix for 1.4 in preparation, that i can publish earliest after 2011-04-21) We'd asked you to retest the issue against a supported branch (Asterisk 1.8), then triage the issue from there. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] sterisk+SS7 Error: chan_dahdi.c: Unable to start PBX on DAHDI/288-1
Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel 2.6.18-194.el5PAE The error is: * * */var/log/asterisk/messages:[Mar 12 07:47:17] WARNING[26421] chan_dahdi.c: Unable to start PBX on DAHDI/26-1* */var/log/asterisk/messages:[Mar 12 07:47:17] WARNING[26421] chan_dahdi.c: Unable to start PBX on CIC 26* Is a very simple PBX which receives the calls in SS7 and redirects them (via IAX2 trunk) to another Asterisk which is connected to an avaya PBX using the same hardware but with PRI singaling. -- Camilo A. Echeverry J. 301 7553789 - Y todo lo que hagáis, hacedlo de corazón, como para el Señor y no para los hombres. Colonences 3:23 -- _ -- 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] How to know how many calls are into hold by asterisk command
Hi All, Is it possible o know how many call are into hold ? who are on hold ? By whom these extension are on hold ? And after 60 sec asterisk will call them automatically as Call Parking do? I wan to make this concept to my PBX system... Thanks in advance -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] How to know how many calls are into hold byasterisk command
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Tuesday, April 19, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to know how many calls are into hold byasterisk command Hi All, Is it possible o know how many call are into hold ? who are on hold ? By whom these extension are on hold ? And after 60 sec asterisk will call them automatically as Call Parking do? I wan to make this concept to my PBX system... Thanks in advance -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer [Danny Nicholas] #1, 2 and 3 can be accomplished using hints and/or AMI. I doubt #4 is possible, but hey I've been wrong plenty of times before. -- _ -- 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] Call Center Reporting
Hi Bilal, Probably there is no open source tool or a good ones available. But few of them I worked with provide up to 2 users free of cost license type of reporting. Reporting for Call Centers can get very complicated. Once you explore some of the commercial apps you will notice how extensive they can get. This is specially true if you are replacing an existing commercial system as you client won't want a mickey mouse replacement but rather a full-fledged call center application. To set down and code for it, it will probably take months to match anything commercially available. I suggest you explore your options before coding it or even attempting queue logs into SQL as that is just he beginning of the work and presentation, ***real-time***, graphs, administration portal, and tons more things are needed to make it a complete suite. Not to forget that this will require continuous updates at the pace of Digium changing Asterisk versions (most of the time as dial-plans changes or queue-log events changes, or if AMI events change). Of course it's possible like other posts suggested but is it economical to embark on it for a single small project? I am not sure Just a thought. -Bruce On Mon, Apr 18, 2011 at 9:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Mon, Apr 18, 2011 at 8:18 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: If all the details you need to compile your reports can be found in existing databases (Asterisk's CDR database stores the details of calls; you may need to get user login/out events from a separate database), Logging the queue_log to MySQL and then setting up a trigger that inserts/updates data to other tables (such as something like agent_status and call_status), along with the CDR, will allow the OP to get pretty much everything they want. (*OP, if you need something substantially more than the stats I mentioned in my earlier post, definitely feel free to email me with details. That way, not only can I help you, but I can make the open source statistics solution I'm working on even better)* A hint: Do the whole thing -- or as much of it as it takes to prove to yourself that you're on the right track -- by hand first, entering all the queries yourself in the mysql prompt (or phpmyadmin), *before* you try to write a program to do it. You will save yourself much heartache that way. AJ, truer words have not been oft spoken! I'd also add that creating views helps if you have complex queries (just to shorten the query that has to be issued from the end program that gets written). -- _ -- 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] [SOLVED] Asterisk thread limit
Solution: The problem is not actually with number of threads.. It is with the stack size. Just reduce stacksize per thread and it allowed thousand of calls :) Also use same configuration to sipp client and server. ulimit -s 1024 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 13 Apr 2011 15:08:15 + Subject: Re: [asterisk-users] Asterisk thread limitver Oops! Asterisk open 419 total thread and stop accepting connection. Can we control number of thread to open or limit ? root@:~# ps -C asterisk -L -o pid,tid,pcpu,state,nlwp,args | wc -l 419 Date: Wed, 13 Apr 2011 10:58:31 -0400 From: lath...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk thread limit On Wed, Apr 13, 2011 at 10:50 AM, satish patel satish...@hotmail.com wrote: Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---[Asterisk][sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r 250 -rp 5000 calls per sec. in this case my asterisk stopped accepting calls at 382 active calls and sipp client through error 1302704824.872674: Can create thread to send RTP packets. (But asterisk is still live to accept calls) I have ulimit is set to unlimited so just wondering is there any asterisk number of thread limitation which we can set to go beyond this boundary? -S Memory limit or load limit might cause this also. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] Linux Based Billing and CDR
I personally know one of the developers of some commercial billing system, supporting what you have written. Not sure for Web site for billing itself, but one of their the projects, using the billing system is www.sip.tj with user panel is at cab.sip.tj All runs Linux. Billing owner can be contacted at bill...@sip.tj . Hope it helps. BR, Vazir 2011/3/25 Jeff LaCoursiere j...@sunfone.com: On Thu, 2011-03-24 at 21:19 +0200, AC wrote: Hi All, Do you'll have any recommendations on a Linux based Customer Management and Pre-paid Billing system for Asterisk, Freeswitch or Kamalio? The system should also allow customers to register, login, buy more credit, view call records, etc. Commercial or Open-source are ok as long as they run on Linux. Thanks, A. -- _ -- 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] T38 fax detection using g729
On 04/14/2011 05:57 AM, Niccolò Belli wrote: Il 14/04/2011 12:25, Larry Moore ha scritto: I made a suggestion on how you could check this i.e. have your incoming call go directly to the fax extension, my 1.8.3.2 installation immediately negotiates a T.38 connection in this sceanrio, of course I enabled the fallback option so it will use g711a if T.38 is not accepted by the peer. Yeah, I know, I just didn't have time to check it. Now I've just finished checking and Eutelia DOESN'T send any T.38 re-invite :( I hoped it was an asterisk bug because there was a chance it may be solved, now that I know it's an Eutelia problem I know it will never be fixed. If you are the receiver of the call (and thus they are the sender of the call), it is *your* system's responsibility to initiate the switch to T.38, not theirs. Note that switching to T.38 and 'faxdetect' are two different, but related, things. You are correct in that if you are using G.729 for the voice path for an incoming call, then detecting the CNG tone from the calling FAX machine will probably not be detected 100% reliably. If you are going to have a DID be able to accept both voice and FAX calls, and discriminate between them based on detection of the CNG tone, then you'll probably have to switch to G.711 a-Law in order to have the most reliable detection. If you are getting free phone numbers, though, you'd be far better off to just assign a dedicated number for incoming FAX calls and not rely on 'faxdetect' at all; this would allow you to use G.729 for your voice calls. -- 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] Softphone IAX
On 19/04/11 1:19 AM, Eduardo Leones wrote: Anyone know a good IAX2 softphone for Windows that has g729 and it is free? That's not going to happen. g729 is not free so how can a softphone that uses it be free unless it isn't honouring the licenses. Don't use this as an excuse to discuss the legality of g729 patents etc - we've been there a million times. Eduardo that last sentence wasn't aimed at you :-) If you're looking for a low bandwidth codec in a sofphone you might consider Speex? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] R: No Internet, no asterisk
On 20/04/11 1:58 AM, Mark Deneen wrote: 2011/4/19 Niccolò Bellidarkbas...@gmail.com: Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Wouldn't a caching nameserver just return NXDOMAIN if it couldn't contact the authoritative server for that domain? It should do - the problem (AFAIK) is that Asterisk is unable to contact the DNS server, not that it doesn't return a result it likes. Therefore a caching nameserver (bind9 etc) should solve it. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] IP Address Management / Open Source / IPAM
Does anyone have a recommendation for an Open Source IP Address Management solution please? There are several commercial players such as BlueCat, BT Diamond, InfoBlox, VitalQIP. And, Solarwinds makes a module that focuses on IPAM. Most vendors tie logic into DNS and DHCP into IPAM designs. In any case, does anyone have awareness of an Open Source solution? Thank you Tom -- _ -- 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] Configure IVR(Inbound and Outbound)
Hi, Is there a step by step guide to Configure IVR(Inbound and Outbound) in AsteriskNow using FreePBX ? Thanks Kaushal -- _ -- 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] No voice in MeetMe for SIP with AGI_BACKGROUND
Hello List, The requirement is little complicated as it is H/W specific. Basically we are integrating a radio gateway (SIP) with asterisk. The gateway will be connected to a meetme room, so that any operator (with IP phone registered as SIP user to asterisk) can login to the room and listen to radio communications and talk. Using a PTT button someone can talk on a radio channel. Once someone presses the PTT button a SIP MESSAGE is sent to the gateway with a string as payload to enable half duplex communication. So, we were planning to run an AGI script with meetme (AGI_BACKGROUND) to receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and to generate a VarSet AMI event. Operator (wants to talk) - SIP:MESSAGE -MeetMe(asterisk)- SIP:MESSAGE - radio gateway And vise versa. Any suggestions on the above scenario. Regards, Rajib Date: Tue, 19 Apr 2011 10:40:05 + (UTC) From: t...@softins.co.uk (Tony Mountifield) Subject: Re: [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND To: asterisk-users@lists.digium.com Message-ID: iojoq5$183$1...@softins.clara.co.uk In article 2658e54b540d284981ea57e6a549ea70abd1f7c...@inblrk77m1msx.in002.siemens.net, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe. The fact that background AGI in meetme only works with Zap channels is a consequence of the original design of Meetme. See these two old posts: http://lists.digium.com/pipermail/asterisk-users/2004-August/050725.html http://lists.digium.com/pipermail/asterisk-users/2004-August/050776.html You will need to change to a different approach to solve your requirement. Could you explain your original requirement? Then people on this list may be able to suggest an alternative way to do it. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. 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
