similar to: Cooperate with SIP ITSP

Hi All, The meetme.conf have created as below: [rooms] conf => 101 conf => 102 and extensions.conf as below: exten => _1XX,1,MeetMe,${EXTEN} why the warning printed when I called 101. WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' for extension (ipcentrex, 101, 1) And I found asterisk have not load the meetme.conf when it starts up.

Hi All, How to park and pickup a call? The scenario of park and pickup described as below. UserA made a call to UserB, and the call ware connected, Then UserB parked (or hold) the call, and told UserC to pickup the call on one line, and then, UserC pressed some keys to pickup the call. Who can tell me what's the Park/Pickup's typical flow in the Asterisk. And how to set the sip.conf,

Guus I have a problem with tinc-1.0pre2-1.i386.rpm This send follow messages Feb 22 18:46:58 enterprise kernel: tap0: Can't register net device feb 22 18:46:58 enterprise tinc.10.1.1.0[6291]: Could not open /dev/tap0: No existe tal dispositivo feb 22 18:46:58 enterprise tinc.10.1.1.0[6291]: Terminating. thanks regards claudio - Tinc: Discussion list about the tinc VPN daemon

================================================================ "Your background and environment is with you for life. No question about that." Sean Connery ================================================================ Release Announcements ===================== This is a maintenance release of the Samba 3.2 series. Major enhancements in 3.2.12 include: o

================================================================ "Your background and environment is with you for life. No question about that." Sean Connery ================================================================ Release Announcements ===================== This is a maintenance release of the Samba 3.2 series. Major enhancements in 3.2.12 include: o

================================================================ "I think there is a world market for maybe five computers." Thomas Watson ================================================================ Release Announcements ===================== This is the last maintenance release of the Samba 3.2 series. Please note that this is the last bugfix release of the

================================================================ "I think there is a world market for maybe five computers." Thomas Watson ================================================================ Release Announcements ===================== This is the last maintenance release of the Samba 3.2 series. Please note that this is the last bugfix release of the

Hey guys, i just wanne report that our debian squeeze xen machines are also influenced by this bug, and we tried the workaround with the debug option by building our own linux-image-2.6.32-5-xen-amd64_2.6.32-31_amd64.deb but this isnt a workaround really, we tested it on an 40 vmachines dom0 and it just dont work, of course the lvm works but the machine is slow as hell and runs into several

Release Announcements ===================== This is the first release candidate of Samba 3.4. This is *not* intended for production environments and is designed for testing purposes only. Please report any defects via the Samba bug reporting system at https://bugzilla.samba.org/. Major enhancements in Samba 3.4.0 include: ------------------------------------------ Configuration changes: o

Release Announcements ===================== This is the first release candidate of Samba 3.4. This is *not* intended for production environments and is designed for testing purposes only. Please report any defects via the Samba bug reporting system at https://bugzilla.samba.org/. Major enhancements in Samba 3.4.0 include: ------------------------------------------ Configuration changes: o

When attempting to call one of the example numbers, like 17474745000, I only get "488 Not Acceptable Here". It works fine when I configure the softphone (Xten X-Lite) to use sipphone's server directly. Am I missing something? Here's my relevant config sections: sip.conf: in [general]: register => 17472442457:mypassword@proxy01.sipphone.com [sipphone] type=friend

I want to build a SER in China, but I don't build a PSTN gateway in China. because goverment don't give permission in China. I want to look for a cooperater, let's my user can dial PSTN to world. if you are interested in my idea,please mail to me. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762

Hi! There. I want to cooperate by translation to Japanese. I was invited to Akemi-san. Already, I translated a few resource of CentOS 5.0. Now translating the CentOS 5.1 release note for Japanese. Best regards. TAIRA Hajime <centos at pantora.net>, web: http://pantora.net/

Hi Ralph, I want to cooperate by translation to Japanese. and I write required information below. AccountName: TadashiJokagi Subject: I want translate Location: my home page, ReleaseNotes, FrontPage(, and others...) -- ----.----1----.----2----.----3----.----4----.----5----.----6----.----7 Tadashi Jokagi mailto:elf at elf.no-ip.org Yokukitana III http://poyo.jp/ Yokukita blog

I'm confused and looking for help For the fist I have Ubuntu x86_64 and I had installed wine 1.1.8 from launchpad and i installed Red Alert 3 when i run it found that no cursor in game, so In order to lay Red Alert 3 with cursos i have compile wine from here git clone git://repo.or.cz/wine/hacks.git then i made symlinks cd hacks Code: mkdir -p `pwd`/lib32 ln -s /usr/lib32/libX11.so.6

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. ?I can

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. I can

Hi, I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a recommendation? Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs... Thanks, Hugh -------------- next part -------------- An