http://blog.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11745 info< at >ipsafari.com 2008-09-06T16:16:39 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11744 On Sat, 2008-09-06 at 13:03 +0530, James Logic wrote: It can be built from source, but the project doesn't currently do RPMs for RedHat. Scott Lawrence 2008-09-06T12:23:52 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11743 James Logic 2008-09-06T07:33:49 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11742 Picher, Michael 2008-09-06T06:51:42 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11741 info< at >ipsafari.com 2008-09-06T01:30:34 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11740 info< at >ipsafari.com 2008-09-06T01:06:39 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11739 Picher, Michael 2008-09-05T23:42:36 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11738 Melcon Moraes 2008-09-05T23:25:05 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11737 Actually, it isn't behind a firewall. I read about the NAT traversal issue and I also read that v4 was going to fix this issue. That's why I decided to move it from my LAN to my WAN. My server has a public IP on it and from what I understand, only passes through a load balancing switch/router, so there is no firewall blocking anything to this machine. I just have no idea what could be wrong. Anyone else have any ideas? ----- Original Message ----- From: Tony Graziano <tgraziano< at >myitdepartment.net> Sent: Fri, 9/5/2008 3:12pm To: Jason King <jKing< at >govdeals.com> ; sipx-users< at >list.sipfoundry.org Subject: Re: [sipx-users] Problems with Voicemail prompts It sounds like you are trying to do NAT traversal with it in its new home. Unless you have something like an Ingate siprator with the proper software modules, this won't work. Version4.0 (current 3.11.x dev version) is working towards this functionality. This is not "stable", but you can yum to it if you download and enable the correct repo for testing. It's different in this version because you can configure the remote traversal and so forth, but be aware it's not ready for production use. If you decide to try and experiment with this, I'd suggest you post question on the sipXbridge module in the sipx-dev list for more assistance. It's good that you got yourself started with the stable ISO version, makes it all less foreign going forward. Good luck. <jKing< at >govdeals.com> 09/05/08 03:25PM >>> I recently installed onto a whitebox for testing. I got internal comm working just fine and voicemail prompts worked too. You could hear the voicemail system giving commands very well. Then I decided to move it outside my LAN and also put it on a nicer server to start fully implementing. The problem I'm having now is that I can call an extension, and if the person is there they can answer and I can hear them just fine, but if they never pickup, or their softphone is offline or not registered, the system throws me immediatly into voicmail (as it should I suppose) but I don't hear anything. The system use to say something like "The person at extesion 513 is unavailable...please leave a message". But now the voicemail just picks up and it shows "connected" on my softphone, but I don't hear anything. I suppose if I waited long enough, I could start leaving a voicemail, but I don't know when to speak because there are no voicemail prompts. I can even dial the operator at 100 (the default autoattendant) and that answers immidiatey, but I don't hear any prompts. It used to say something like "You have reached an automatted answer system...etc.etc" Now it says nothing when it answers. I know you can dial and extension while in that auto attendant to be transferred, which I can still do, but the system isn't prompting me to do anything. I just dial the extension, then I'm transferred, and if that extensions voicemail picks up, again, I get no prompts to do anything. There is no automatted attendant/voicemail prompts at all. This is a fresh install of sipXecs 3.10.2 on a Dell PowerEdge 1850 server. The install is from the install CD, I didn't compile this from source or install it into a certain distribution. Thanks, Jason King Senior Network Engineer GovDeals, Inc 334-387-0467 Jason King Senior Network Engineer GovDeals, Inc 334-387-0467 _______________________________________________ sipx-users mailing list sipx-users< at >list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users Jason King 2008-09-05T22:55:43 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11736 Tony Graziano 2008-09-05T20:12:24 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11735 Jason King 2008-09-05T19:25:35 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11734 Actually only 104 is not supported ( obsolete ) although 114 is. I'll just configure it using its web interface. Its just for testing anyway. Ranga M. Ranganathan 2008-09-05T19:01:57 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11733 Tony Graziano 2008-09-05T15:56:06 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11732 If you want sipxpbx to be recognized as a service (e.g., so /sbin/service can manipulate it and so it starts upon boot), you can copy (or link) .../etc/init.d/sipxpbx into /etc/init.d/sipxpbx, then use chkconfig as usual. Dale Dale Worley 2008-09-05T15:36:55 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11731 You can disable voicemail for one user, and I think that you can adjust how long the phone rings, but you can't set sipX to make the call ring truly forever. Indeed, I think the SIP protocol has a limit of 3 minutes. Dale Dale Worley 2008-09-05T15:34:20 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11730 I think it is supported: http://sipx-wiki.calivia.com/index.php/HowTo_configure_AudioCodes_SIP_Gateway_with_sipX#FXS_Gateways Damian Krzeminski 2008-09-05T15:08:43 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11729 Hello, Does Audocodes FXS MP 104 gateway work with sipx? It is not supported by sipxconfig. I suppose one would need to configure it through its own web interface. I am trying to set one up for interwroking experiments with sipxbridge. Thank you in advance. Regards, Ranga M. Ranganathan 2008-09-05T14:37:51 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11728 On Fri, 2008-09-05 at 14:09 +0200, Kerker Staffan wrote: Yes, it's getting attention. Whether or not it will be a "supported" feature in 4.0 is still TBD. In particular, TLS for ITSP connections (which, as I recall, was your big interest) is a possibility. If you're able to help test this, let Ranga know. Scott Lawrence 2008-09-05T13:27:07 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11727 Hi I m stuck somewhere while building from svn unstable source: When i run service sipxpbx start I got: sipXpbx: sipXpbx configuration problems found: sipXpbx: sipXpbx: Check Apache configuration sipXpbx: Syntax error on line 301 of /usr/local/sipx/etc/sipxpbx/httpd.conf: sipXpbx: Error:\tApache has not been designed to serve pages while\n\trunning as root. There are known race conditions that\n\twill allow any local user to read any file on the system.\n\tIf you still desire to serve pages as root then\n\tadd -DBIG_SECURITY_HOLE to the CFLAGS env variable\n\tand then rebuild the server.\n\tIt is strongly suggested that you instead modify the User\n\tdirective in your httpd.conf file to list a non-root\n\tuser.\n sipXpbx: sipXpbx: Check sipxcallresolver sipXpbx: Checking for running sipXproxy... not running (ok) sipXpbx: psql: could not connect to server: No such file or directory sipXpbx: Is the server running locally and accepting sipXpbx: connections on Unix domain socket "/tmp/.s.PGSQL.5432"? sipXpbx: Error: PostgreSQL is not running. sipXpbx: Checking configuration... sipXpbx: proxy configured for logging sipXpbx: call resolver configuration ok sipXpbx: /usr/local/sipx/lib/ruby/gems/1.8/gems/sipxcallresolver-2.0.0/lib/main.rb... ok sipXpbx: sipXpbx: Check sipxconfig sipXpbx: psql: could not connect to server: No such file or directory sipXpbx: Is the server running locally and accepting sipXpbx: connections on Unix domain socket "/tmp/.s.PGSQL.5432"? sipXpbx: Postgres configuration database not detected. sipXpbx: Run (as root:) sipXpbx: sipXpbx: /usr/local/sipx/bin/sipxconfig.sh --setup sipXpbx: sipXpbx: Check sipxivr sipXpbx: ==== /usr/local/sipx/etc/sipxpbx/autoattendants.xml: sipXpbx: Not readable. sipXpbx: ==== /usr/local/sipx/etc/sipxpbx/validusers.xml: sipXpbx: Not readable. sipXpbx: Attempting to start despite configuration problems Starting sipXpbx: Starting sipxsupervisor: [ OK ] Starting httpd: Syntax error on line 301 of /usr/local/sipx/etc/sipxpbx/httpd.conf: Error:\tApache has not been designed to serve pages while\n\trunning as root. There are known race conditions that\n\twill allow any local user to read any file on the system.\n\tIf you still desire to serve pages as root then\n\tadd -DBIG_SECURITY_HOLE to the CFLAGS env variable\n\tand then rebuild the server.\n\tIt is strongly suggested that you instead modify the User\n\tdirective in your httpd.conf file to list a non-root\n\tuser.\n Please specify how to resolve it. Thanks, Vikas VG 2008-09-05T13:15:23 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11726 Kerker Staffan 2008-09-05T12:09:25 http://permalink.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/11725 Dean Hiller 2008-09-05T09:23:21 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.sipfoundry.general