search for: registerd

...r at mushroom ~]$ Does anyone out there have some magic incantations to make Yum re-install the sound system? CentOS release 6.3 (Final) Linux mushroom.patch 2.6.32-279.11.1.el6.i686 #1 SMP Tue Oct 16 14:40:53 UTC 2012 i686 i686 i386 GNU/Linux -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 https://linuxcounter.net/ ****

...one, I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_translator_activate I have tried i686, i386, athlon, and athlon-xp versions of the codec but none of them have loaded. Any help wou...

Hi I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then go to the VOIP serviece provider network ip-phone ->asterisk->cisco->voip netwrok how can i do that , whats asterisk config file ( H.323, sip and extensions.conf ) and how...

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The problem is that the SIP user receives the call with this uri: <sip:xxx at 10.10.10.2> The call work...

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks...

Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans

By chance, has there been any progress on a 5.0 version for an Ultra Sparc? -- Best regards, Chris Registerd Linux user number 448639

Hey Y'all, why am I getting double copies of every email on this list today when it wasn't happening yesterday? Isn't happening on any of my other email. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004

...now where I can find source packages for X86-64 CentOS 6.2: dev86 is needed by my project iasl is needed by my project spice-server-devel >= 0.8.2-4.el6 is needed by my project I can't seem to find them in the source repositories. Thanks. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 www.counter.li.org ****

Hey Y'all, What application replaces the functionality of Rhythmbox for Cent OS 6.2? In particular I am interested in handling podcast feeds. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 www.counter.li.org ****

...ot;x246-0.120-5.20120303.el6 (i686)" available in my Add/Remove Software tool. I can't find said package in any of the repos I've searched through. Can anyone point me to a link where I can download that package to my local repo? -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 https://linuxcounter.net/ ****

...gainst the extension.. But my exact requirement is like this 1)Call to the user 2)if answers connect him to the extension provided in the extensions.conf 3)if the user didnt lift the phone within the deault timeout period(30 sec) 4)if the user cancels the phone (Congestion case) 5)if the user not registerd to the(unreachable case) to trace the cases of 3, 4, 5 how should i follow the API I got confused with originate action,orginate sucess event , originate failure event can anybody give me a hint so that i can proceed further thanks in advance for the kind suggestions. regards srinivas antarved...

...o know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on. At last, my need is: Accept a non-registerd sip-strem from a well known ip address (and only from that ip address....) I tried to add a [testsip] ;username=testsip type=friend ;secret=testsip qualify=no port=5060 nat=no host=x.y.z.w dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="test sip " <testsip> that...

...happy. Anyone know how to find the Tcl library location, Tcl header file location, and Tcl library base name? CentOS release 6.3 (Final) Linux mushroom.patch 2.6.32-279.19.1.el6.i686 #1 SMP Wed Dec 19 04:30:58 UTC 2012 i686 i686 i386 GNU/Linux -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 https://linuxcounter.net/ ****

...est software update? 2. What course of action would you recommend to restore the operation of openoffice.org? CentOS release 6.2 (Final) Linux mushroom.patch 2.6.32-220.13.1.el6.i686 #1 SMP Tue Apr 17 22:09:08 BST 2012 i686 i686 i386 GNU/Linux -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 www.counter.li.org ****

...ecieve invite message and it forwards to asterisk 1)how the Asterisk will handle this call with rtp 2)and when pstn customer calls the call goes in to SER and it looks the 'location' database and it will reject call because it is not registerd user so, we take pstn call directly to asterisk and we forward call from asterisk to SER and i want to know is how the SER handle this call that means when SER found a sip client it invites that sip client and which mediaproxy is going to handle this call the SER's or A...

...he stellarium rpm is in but I do know that it's out of date. Who's cage should I rattle to get the repo up to speed? CentOS 6.2 Linux mushroom.patch 2.6.32-220.4.2.el6.i686 #1 SMP Tue Feb 14 00:24:28 GMT 2012 i686 i686 i386 GNU/Linux -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004

...o : capture 1 02-00: USB Audio : USB Audio : playback 1 : capture 1 I tested the mic by plugging it into my wife's XP machine. The mic works fine but the XP machine doesn't. Very frustrated. Anyone have any idea what else I can check? -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004

...hingWell.wmv . [mlapier at mushroom Media]$ [mlapier at mushroom Media]$ gst-typefind-0.10 WishingWell.wmv WishingWell.wmv - video/x-ms-asf [mlapier at mushroom Media]$ Do you need to see the output of gst-feedback-0.10? It's quite lengthy. -- _ ?v? /(_)\ ^ ^ Mark LaPierre Registerd Linux user No #267004 https://linuxcounter.net/ ****

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper = 62.225.189.250 AllowGKRouted = yes context=local ; [time] type=...