[asterisk-users] g.729 on solaris10/x86
Hello, I'm looking for a way to have G.729 codec working on Solaris/x86. Binary codec from Digium is not compiled for Solaris/x86 (only sparc). Are there any alternative (free or commercial) G.729 implementations, which would work? I saw something from Intel and got it to compile on Linux, but it was only for evaluation purposes, so we upgraded to commercial codec from Digium. I really don't care about the U.S. patent, it does not apply here, only about copyright. If there's something with source code (could be commercial), that I can make work on Solaris, it would be great. Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 debian packages
Hello, are there any (possibly experimental) asterisk debian packages (or at least a debian/ directory to build our own)? Previously I used to modify debian/ directory from earlier version, but 1.4 changed build process, so this is not that easy. Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lots of registrations, sip problem
=5060..From: sip:4212326601 [EMAIL PROTECTED];tag=as6550d352..To: sip:[EMAIL PROTECTED];tag=as12748c70..Call-ID: [EMAIL PROTECTED] 7.67.16.43..CSeq: 42620 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ..Contact: sip:[EMAIL PROTECTED]..WWW-Authenticate: Digest realm=provider, nonce=311081ba..Content-Length: 0 Until (**) it seems like a standard registration. Everything after that is repeated several times a second. I don't know why it tries to register, when it has clearly already done so. Even sip show registry shows that provider as registered. Please note the difference in URI, first time it's sip1.provider.com, second time it's sip:my-provider-link. This is my sip.conf (relevant parts of it): register = 4221917293125:[EMAIL PROTECTED]/4221917293125 [my-provider-link] type=friend username=4221917293125 secret=secret123 host=sip1.provider.com nat=no context=provider-in canreinvite=no insecure=very qualify=no It seems to me, that it tries to register also this my-provider-link somehow, but I have no idea how and why. Just for the record, I have several accounts of this SIP provider in my sip.conf (about 10). All other providers are working correctly, this one (except for these excessive registrations) is working too (all of the accounts). I've been told by my voice provider, that they are also using Asterisk on their side. I've tried upgrading from Asterisk 1.2.10 to Asterisk 1.2.12.1 and it did not help. Any ideas or help would be greatly appreciated. Juraj Bednar. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quality control
Hello, I would like to create some form of reporting of call quality. Is there a way to collect quality of RTP data (for SIP calls) to gather some statistics (packet loss, ...). I would like to know when calls are of lower quality and if I should blame ISP, operator or look for some problems on my setup. Thanks, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] weird sound with IAX
Hello, I am having very weird sound on IAX protocol (using SIP, it seems to work OK). I use Asterisk 1.2.10. As a client, I use Idefisk. Today, i let two completely different asterisk machines talk to each other, with more or less same results. I currently do not use IAX trunking. This test was performed on 1Gbps ethernet with no packet loss with ulaw codec (no transcoding on the way): http://flz.sk.cx/audio/20060831-181241_59206988_to_.wav.mp3 Any help would be greatly appreciated. I looked at voip-info.org, but saw no troubleshooting info for IAX. Thanks, Juraj Bednar. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk presence (from manager API)
Hello, Did you try a combination of qualify=yes in sip.conf and then try the ExtensionState in the manager? yes, I have qualify=yes in the IAX config for peers I'm interested in. Seems like if qualify=yes or 2000... whatever, is not set then asterisk will not always know the state of the phone if it looses registration. That would seem to explain the problem you have with extensionstate. I can set qualify=2000, currently I have qualify=yes. Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk presence (from manager API)
Hello, I would like to somehow get the presence of IAX2 and SIP users from Asterisk Manager API or using any other means. I tried watching for PeerStatus event, but it seems unrealiable (http://bugs.digium.com/view.php?id=7833). I tried defining hint for user and sending ExtensionState event, which is also unreliable (once I had qualify OK status in iax2 show peers, I could receive calls and I got status of 4, which is unavailable). How to get reliable information about peer status? I have qualify=yes in all iax friends, I am using realtime and I can receive calls or dial without any problems. Thanks, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk presence (from manager API)
Hello, Google is your friend: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState not today. I mentioned in my original mail, that ExtensionState is unrealiable too. Sometimes I quit my softphone and I see extension as Idle (status 0), sometimes I log in and the extension is shown as unavailable. Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GROUP() and queues
Hello,I have a call queue with ringall strategy. Users are IAX2 users. I would like to allow only one parallel call at all. I tried setting incominglimit=1 in iax.conf, but this did not help. I want queue to ring only when operator is not on line already with someone.
I tried creating Local channel for operators. I added members = Local/[EMAIL PROTECTED]members = Local/[EMAIL PROTECTED]
to my test queue. Then I defined this in my extensions.conf:[agents_test]exten = _9.,1,Set(GROUP()=${EXTEN})exten = _9.,2,NoOP(Group count is ${GROUP_COUNT()}, group is ${GROUP()}, exten is ${EXTEN})
exten = _9.,3,Dial(IAX2/user${EXTEN},120,rtT)exten = _9.,4,Goto(exithandling,s-${DIALSTATUS},1) Now, when I call queue first time, both of agents (998 and 999) are ringing. When the call is answered, I dial second time, but both channels' NoOP results in group count 1. So my question is how to make it work. What am I doing wrong? Or is there a simpler solution? Is there a reason why incominglimit=1 in
iax.conf does not work? I have asterisk 1.2.7 (with security patches). Any help appreciated. Thanks, Juraj Bednar.
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[Asterisk-Users] transfer outside of a call?
Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom soundpoint ip600 problem
Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk console shows me warning chan_sip.c:3250 process_sdp: Error in codec string 'eo 0 sip 34 103'. Running CVS Head, some older build. Any ideas what could be wrong will be very helpful. Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian sarge package for 1.2beta1?
Hello, has anyone seen or created a Debian Sarge package for 1.2beta1? I saw some for Sid, but I prefer not upgrading glibc right now, as this is a production server (Asterisk on it will be for testing). Thanks, Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voice Encryption
Hello, I went over the code. AES128 is the only algorithm that is suppored today. More importantly there are some concerns on the vulnerability as discussed in http://lists.digium.com/pipermail/asterisk-security/2005-August/60.html. People are using UDP VPNs to satisfy customer requirements. http://lists.digium.com/pipermail/asterisk-users/2005-August/120293.html we are using plain ipsec here. From softphones, we just use operating system's native ipsec support. For hardphones, we have a custom device based on soekris board with vpn hardware encryption accelerator, which does the job for the phone. Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Eyebeam
Hello, What's the status on using eyebeam with Asterisk, does it still require a patch to Asterisk to support the video component? I'm intererested in starting to use Video and audio telephony but wary of anything that requires patches. cvs head works out of the box, just enable the h.323+ codec. cvs head is what will become asterisk 1.2, check for betas... Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxcomm huge latency
Hello, I use iaxcomm-latest from the iaxclient.sf.net page (binary release) on linux, also tried Mac OS X version with the same result and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of seconds, constantly changing over time. It was run on two different machines, always to a SIP phone (which otherwise works correctly even with VoipBuster, which also uses IAX with no latency and other SIP phones). Is it a known bug? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip messaging (tested on eyeBeam) support
Hello, I added queuing support (based on SQLite database to store the queue) for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs, but it's at least something. I created page about installation on the tips and tricks of voip-info.org: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+Messaging Any bugfixes are welcome. Yes, it's a huge hack and supports only sip to sip messaging based on presence hints. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
Hello, I just got my Soekris 4801 box for use with Asterisk, but not as a primary Asterisk server. * [EMAIL PROTECTED] (Is @home or regular better?) If you want to run from CF, I recommend running some distribution (that does not take much space) and your own Asterisk... I'm not even sure if it be that easy to install Asterisk on Soekris in the first place. I found documentation not being that good for installs, I found a wonderful page describing the install process: http://www.ultradesic.com/index.php?section=22 * Shorwall firewall Try to get a real firewall, Shorewall has quite high latency. You should optimize... * QoS This could be quite CPU intensive... * astcc * h.323 module * wakeup * festival (Maybe the CPU / RAM is too low for that) * MOH * voice mail * ??? Some things are quite CPU intensive. Why do you want to use Soekris BTW? Some more powerful Mini-ITX could do the job better I believe. Don't take me wrong, I love those Soekris boxes, but it's 266MHz Goede processor, you want far too much from it I believe... Some questions about Soekris: What is in the package? (Power adapter?, CF?, manual? ...) Nothing, just box and the board. How to install it? See the link above. What is the CF size you are using? and how much is still free? What have you installed? For my setup I installed OpenBSD, although I primarily use Debian GNU/Linux. The OpenBSD choice was because of the vpn card for Soekris, which is better supported under OpenBSD. I installed the base package except games and manual pages, about 60MB was still free (I used 256MB compact flash card). Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP messengers video phones
Hello, There's some work on creating a multiprotocol solution for instant messaging within Asterisk, but it will not be in the coming v1.2. is the work somewhere as a patch to be tried or in some other form, even if it's not coming to 1.2? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence in cvs head - how does one map extension to sip user?
Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client subscribes to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence in cvs head - how does one map extension to sip user?
Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client subscribes to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Yes, there are. Check the hint priority in your extensions.conf.sample in the source directory. Basically you connect an extension to one or several devices by entering a hint: exten = 500,hint,SIP/juraj /Olle again, thank you very much for explaining this. I added this piece of information to the voip-info.org wiki, as many people have been asking this on -users list before and there was a lack of information. If anything on the page is not correct, feel free to edit: http://www.voip-info.org/tiki-index.php?page=Asterisk+presence Best wishes, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Interface with mobile phone
Hello, There's this device called VoiceBlue GSM gateway. It talks gsm on one side and SIP on the other side. Have a look at: http://www.voip-info.org/tiki-print.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX yep, but it is very expensive, I found. Even cellphone + cellsocket + FXO card would be cheaper than this. I want to do the same thing, I will try to use chan_bluetooth, but its' svn repository is unaccessible right now :(. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g.729 codec -- open source?
Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and IM again, want to develop a working hack
Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and found lots of wonderful proposals, which try to convince asking users, what needs to be done to support this well (multichannel, multiple phones per user, ...), mainly saying, that without very difficult reworking of internals, it would not be supported. What I really need is to hack it into chan_sip.c. I need the support of other channels and applications (f.e. MeetMe), but where I really care about presence and IM is SIP. So, any project, hack, patch, anything, that would allow me to go further with this would be greatly appreciated. I found this page in Russian: http://www.asterisk-support.ru/forums/development/53843189454 that somehow deals with the problem. I tried babelfish translation, (http://babelfish.altavista.com/babelfish/trurl_pagecontent?lp=ru_entrurl=http%3a%2f%2fwww.asterisk-support.ru%2fforums%2fdevelopment%2f53843189454) but I was not able to find out, if it really at least partially solves this problem, but as far as I understand it, Windows Messanger makes use of Subscribe/Notify, so this should be it. b.) Anyone has a partial solution using SER (which supports presence and IM) as a frontend, but routing all calls through Asterisk? Can this be done? I need the calls to go via Asterisk (I don't mean only sip notifications, but also the data, so I have canreinvite=no). So basically, SER would be a registrar proxy to Asterisk, which would do the authentication. The only thing, that SER would do would be to handle presence and IM and pass everything else on to Asterisk (as far as I know, SER can't pass traffic through it. I need the data to pass through the SIP server, since machines in my network topology don't see each other, it's a star with Asterisk in centre -- quite poetic indeed:). Any ideas, pointers to similiar configurations, ... are welcome. c.) If there is no solution to start with, is it possible to implement it only to chan_sip? I'm not familiar with Asterisk source code at all. Where are the places to look (in chan_sip.c) which are best to hook this code. Again, any code, hints, etc. about the structure of the source code are really welcome. Doing this in a clean way (although it's a hack) so it can be reused by community as much as possible is my intent. If anyone wants to help with the project by donating coder's time, mail me off the list. I hope I'll be able to support presence for hardphones and Xten's eyeBeam softphone in a few days with your help. Best wishes and thanks for any replies, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bluetooth audio and asterisk
Hello, Has anyone successfully used a standard bluetooth enabled system to connect to a standard bluetooth enabled mobile phone (not the bluetooth to FXS converters) to create an audio path for phone calls with asterisk, if so is there a writeup on what was done so that others can replicate this. I want to do the same thing. I want to buy an bluetooth capable mobile phone without a dongle. I want asterisk to predict it's a headset, so it can route audio. This way, I could accept calls from mobile network for very cheap. And also do the opossite route (call to cell-phone network). Check this: http://www.voip-info.org/tiki-index.php?page=Asterisk+Bluetooth+channels If you'll have any progress in this area, please let me know. It should also be able to send SMS, I have previously done this successfully, but without Asterisk using smstools. For sending sms, using a web gateway would be probably cheaper, but it should be good for receiving. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and fayn.cz
Hello, I would like to use Asterisk with fayn.cz service. They should be using a standard H.323 protocol, but I have no more information about this. They provide a softphone and/or rebranded H.323 telephone, but I don't know any H.323 settings nor if the firmware in the phone is modified. Has anyone tried this successfully? They provide a Prague telephone number reachable from classic telephone network in their free monthly plan, which would be of great use for me, since I have many friends in Czech Republic and it would be very cheap to call to a local prague number instead of international call. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi IP Phones
Hello, I know there are wifi sip phones out there but I have a question, are any of these phones anti explosive? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. BTW: it would probably be even cheaper to use a normal analogue wireless phone and connect it's base station to FXO port. You can get cordless analogue telephones for cheap these days and I believe when you buy a card with FXO to connect the base station to them, it would be even cheaper than buying a Wifi phone... Have you considered this? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gsm gateway hardware recommendation?
Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also sending and receiving SMS will be a plus. I have a friend living in luxembourg, which would like a slovak phone number to communicate with friends. It would end on my server at home and all calls to his sim card will be routed to his ip telephone in luxembourg (and vice versa). Support for more than one sim card is a plus. Since it's a home/hobby use, I would prefer a low-cost solution. Any ideas (may be off-list) are welcome). Thanks, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and video conference
Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
