similar to: [SIP/H.264] Codec negotiation problem ?

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

We have here a TandBerg "videoconferencing system" connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtaining audio calls, the video not appears. When i try to make a 64K calls (data calls), the call does not

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this -

Hi, For some time now (starting with the 3.10 or 3.11 kernels, but I'm not sure) I have the following problem: I have a machine with 6 SATA slots and two SAS controllers, one onboard HBA and one RAID controller in a PCIe slot. The problem is that the order of the SAS controllers changes randomly after a reboot, so the SCSI addresses of the devices change. One time lsscsi prints (empty

Friends, SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

hi, If i try a backup with tar / tape sdlt320 on a debian-sarge / 2.6.12 based XEN 3 System, the System freeze. If i test this by knoppix the backup works fine. Test with xen 2.07, Kernel 2.6.11.12 on xen 3.0.0, Kernel 2.6.12 This System in a Interserver with an LSI scsi-Controller and a Tandberg SDLT320 Streamer. Are thereany known issues with this config and xen ? regards Rene

Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4

Hi, I'm trying to configure Asterisk as SIP-H.323 Gateway via ooh323, but I have an error relatively to the GK Confirmation message. >From the log: "H323 RAS channel creation - succesful Sent GRQ message Gatekeeper Confirmed (GCF) message received ERROR:No Gatekeeper ID present in received GKconfirmed message Ignoring message and will retransmit GRQ after timeout Error: Failed to

For the last 10 years we have had some Tandberg video conferencing units spread across our WAN. To make a long story short we had to work with another entity to allow access into those units with H.323. Calls from public networks have never worked to those devices with private IP's. Over the summer, my coworker and I built an * VOIP system for one of our high schools. The need came up for

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

Just thought I'd remind you about my problem ;o) Has anyone got a clue about what I need to do to solve the wine/java problem below? - Anders Anders R?nningen wrote: > Dan Kegel wrote: > >> anders.ronningen@tandberg.net wrote: >> >>> I am using wine to compile code for Texas Instruments dsp's. One step >>> in the process is generating code from

Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type:

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Hi I am having a real nightmare over this: Setup is : win2k as ADS server linux samba server is: # smbd --version Version 3.0.14a-Ubuntu kerberos: # klist Ticket cache: FILE:/tmp/krb5cc_0 Default principal: Administrator@KCS.COM Valid starting Expires Service principal 07/03/06 09:51:51 07/03/06 19:51:51 krbtgt/KCS.COM@KCS.COM Kerberos 4 ticket cache: /tmp/tkt0 klist: You

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all. I have 15 or so CentOS servers (now v4.4, I will upgrade them all to 5 when it comes out) and a few windows machines. I am looking to implement a network backup solution using Amanda. I will be backing up 200 gig or so to start, but that will grow. Does anyone have any recommendations on tape drives that will work "out of the box" with CentOS? Thanks, Joe