search for: krief

Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone 877-777-2932

Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ? For instance: Location 1: - 200 users on a legacy PBX - among those users, 50 have access to voicemail service - TDM trunk to Location 2 Location 2: - 100 users on Asterisk - PSTN access - TDM

OK... maybe I got a little anxious and ran out and bought a Tyan GX28 with dual Opteron (dual core) processors. (It is a nice server ;) ) I did neglect to find out that you can not manually set the IRQ's on this motherboard. I am now stuck sharing an IRQ with the ethernet controller and no foreseeable end to my dilemma. I have a Digium TE210P and zttest consistently runs at 99.97% which

...You have two different numbers, your mobile number and your IP number And these cant automaticly be transferred. Hope this answeres your question Regards Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Sch?pzinsky <HYPERLINK "mailto:jos@detele.dk"jos@detele.dk>: Hello Be aware that the Nokia...

...er And these cant automaticly be transferred. Hope this answeres your question Regards Jon _____ Fra: asterisk-users-bounces@lists.digium.com [HYPERLINK "mailto:asterisk-users-bounces@lists.digium.com"mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Sch?pzinsky <HYPERLINK "mailto:jos@detele.dk"jos@detele.dk>: Hello Be aware that the Nokia...

Hello, I tried to read Bristuff source code to understand the way calls are bridged from one BRI port to another (as HFC cards have "active channel switching capability"). Doing so I looked at zaphfc.c file which seems to be the only .c file constituting bristuff but I may be wrong I'm far from being an expert on this. I found 4 functions (hfc_btrans, hfc_brec, hfc_dtrans,

...------------------------- * ASTERISK-28267 - res_stasis: Add ability to switch applications (Reported by Benjamin Keith Ford) Bugs fixed in this release: ----------------------------------- * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate (Reported by Olivier Krief) * ASTERISK-28350 - manager: Stasis backed up due to locking (Reported by Joshua C. Colp) * ASTERISK-25792 - chan_sip: qualifygap bounds checking (Reported by Paul Sandys) * ASTERISK-28341 - res_config_odbc eliminates empty custom (���@��� prefix) variables (Re...

Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol

...does not cause hangup (Reported by Joshua Colp) * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash (Reported by Sean Bright) * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed (Reported by Olivier Krief) * ASTERISK-25637 - Multi homed server using wrong IP (Reported by Daniel Journo) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25640 - pbx: Deadlock on features reload and state change hint. (Reporte...

...4 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28759 - A non n...

...4 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28759 - A non n...

...ase: ----------------------------------- * ASTERISK-27541 - app_queue: Queue paused reason was (big number) secs ago when reason is set (Reported by C��sar Benjam��n Garc��a Mart��nez) * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate (Reported by Olivier Krief) * ASTERISK-28350 - manager: Stasis backed up due to locking (Reported by Joshua C. Colp) * ASTERISK-25792 - chan_sip: qualifygap bounds checking (Reported by Paul Sandys) * ASTERISK-28341 - res_config_odbc eliminates empty custom (���@��� prefix) variables (Re...

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

...4 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Ha...

...4 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Ha...

...4 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Ha...

...when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) * ASTERISK-26717 - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) * ASTERISK-26643 - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) * ASTERISK-25237 - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) * ASTERISK-26857 - chan_pjs...

...when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) * ASTERISK-26717 - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) * ASTERISK-26643 - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) * ASTERISK-25237 - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) * ASTERISK-26857 - chan_pjs...

...does not cause hangup (Reported by Joshua Colp) * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash (Reported by Sean Bright) * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed (Reported by Olivier Krief) * ASTERISK-25637 - Multi homed server using wrong IP (Reported by Daniel Journo) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25640 - pbx: Deadlock on features reload and state change hint. (Reporte...

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would