Re: [asterisk-users] Voicemail maintenance
Jordan Novak wrote: Has anyone created a GUI for this. I am not sure what youre looking for but we developed a Voicemail Manager: = http://sip-syndication.com best regards, Arnd ___ --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 gui sans live cd
Patrick Aljord wrote: is there a good and free asterisk gui that is not tight to a live cd? I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I just want to run asterisk on my debian install. Is there a way to run [EMAIL PROTECTED] on debian? or anything similar? You can install freepbx ([EMAIL PROTECTED]) on any linux box. Wheres the prob?! ___ --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] New Version of Tycho Voicemail Manager released
Hi, we are releasing an update of our Tycho Voicemail Manager. The update to Beta 0.2 contains a bugfix and a couple of improvements over the 0.1 version: Bug fix: * missing Channel Type added to extension subscription Improvements: * adjustable refresh interval (voicemail) * manual refresh button (voicemail) * re-open windows after application restart (extensions, voicemail) * Support Forum menu item - Web Link to our support Forum * Reset Perspective menu item - Resets the windows to it's defaut locations Please note that this version will delete your preference settings as well as any defined voicemailboxes and extensions. Help us and report all problems/bugs to our support forum. Thank you. Note for Trixbox Users: When subcribing to an extension please use the context from-internal in the context field of the subscription menu. For more Info, documentation and Downloads go here: http://sip-syndication.com best regards, Arnd ___ --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] Anyone using Voicemail with IMAP Support?
Hi, ive tried to setup a svn trunk version of asterisk to test voicemail with imap support and i am so far without success. Is there _anyone_ running voicemail with IMAP Support who can answer some basic questions? regards, Arnd ___ --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] voicemail access thru apache on another server
Hi Benjamin, Am trying to build a system, wherein users can access their profiles, and hence voicemails thru a browser. I am using Apache and am running it on another box and asterisk on another. Am keeping them seperate to not have http traffic on the same box as asterisk. Now, my qs: Is there a way to tell Asterisk to store the msg.txt information in an sql database, so that it's easier to access the voice mail info?? I would wait until the IMAP support in asterisk (currently only in the svn version) is stable instead using sql-db based storage. At least if were talking about an medium installation or bigger. Also, any way to run a script or something, to move a message from INBOX to Old, when a user listens to the message thru the web browser?? Now, how on earth do i read the recordings and play them out thru a browser!! I did write a proof of concept script in php for accessing and manipulating the voicemail folder. It has no locking atm therefore there are some race conditions and its not usuable in a large scale production environment. Youre welcome to add session/locking suport though :} The script enables you to view, read, move, delete and forward voicemails using URLs. ascii based and asterisk realtime authentication is supported. You can use it at least to extract the code how to send an audiofile (if youre using php that is). Should be no problem to convert the scripts to perl or any other script language. Script: http://sip-syndication.com/index.php?option=com_remositoryItemid=26func=selectid=2 Documentation: http://sip-syndication.com/index.php?option=com_contenttask=categorysectionid=5id=21Itemid=47 cheers, Arnd ___ --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] Setting up imap based voicemail / invalid remote specification
Hi,
ive just installed a svn trunk (r42858) and i am having problems
getting app_voicemail to even try to connect to a imap server.
Ive added the following to voicemail.conf
--
; new IMAP Stuff
imapserver=mydom.com
imapport=143
expungeonhangup=no
[..]
[default]
; Office Accounts
7709810 = 1234, Me Myself, [EMAIL
PROTECTED],,attach=yes|imapuser=me|authuser=me
|autpassword=mypass
--
When trying to leave a voicemail ill get:
--
[Sep 13 16:20:45] ERROR[30445]: app_voicemail.c:8193 mm_log: IMAP Error: Can't
open mailbox {mydom.com:143/imap//user=me}INBOX: invalid remote specification
[Sep 13 16:20:45] ERROR[30445]: app_voicemail.c:2455 count_messages_imap:
Houston we have a problem - IMAP mailstream is NULL
[Sep 13 16:20:45] NOTICE[30445]: app_voicemail.c:2876 leave_voicemail: Can not
leave voicemail, unable to count messages
--
Any hints?
BTW: If i call Voicemail with Voicemail([EMAIL PROTECTED]|sb) in
extension.conf the ${CONTEXT} gets replaced with the actual context but
asterisk still tries to find the mailbox in the default context? Am i
missing something here (Docu for changed extension.conf syntax? = where?)
or is this a bug?
regards,
Arnd
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[asterisk-users] Anyone working on VXML, CCXML support for asterisk?
Hi, is there anyone working on VXML or CCXML integration for asterisk? If not, anyone interested in developing it? -- Arnd ___ --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] Anyone working on VXML, CCXML support for asterisk?
Hi, Asterisk Mail List wrote: I've integrated OpenVXI 3.4 (the latest one) with Asterisk for a client. It is now in production, interpreting their VXML pages using Asterisk for SIP/IAX telephony [..] I also plan to release the code under the GPL as soon as I can figure out the best way to do it. i would like to test this if possible. Would be cool if you could send me an email in case you release it to open source. I'll be at the VON developers and users group meetings today. Unluckily i am not at the VON meeting this year. regards, Arnd ___ --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] SER+Asterisk integration
have a look at the nathelper examples in SER distribution. This is from
an rather old installation of mine.
--
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test(3)) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == REGISTER || ! search(^Record-Route:)) {
xlog(L_ERR, LOG: Someone trying to register from private
IP, rewriting\n);
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority i
s
# smart enough to be symmetric. In some phones it takes a co
nfiguration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with
kphone it is
# called symmetric media and symmetric signalling.
fix_nated_contact(); # Rewrite contact with source IP of sig
nalling
if (method == INVITE) {
fix_nated_sdp(1); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6);# Mark as NATed
};
};
..
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
--
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Re: [asterisk-users] Asterisk as a SER client
Andrea Spadaccini wrote: are there any particolar guidelines to follow in order to make Asterisk act as a SER client? No. I have the following config: register = account:[EMAIL PROTECTED]/asterisk-extension and [ser-out] type=peer secret=fump host=serbox.com callerid=MyMyselfAndi 123456 username=account fromuser=account fromdomain=serbox.com context=yourcontext insecure=very Canreinvite=no nat=no qualify=no ___ --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] SER+iptables+Asterisk
Siqhamo Sifo wrote: I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . Use rtproxy for SER and an according ser.cfg (see SER example configs) -- Arnd ___ --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] Voicemail/Email Integration
[EMAIL PROTECTED] wrote: Is there a way to implement voicemail/email integration such that you could retrieve the voicemail with either the phone or email, but only have to delete the message once? You can try our voicemail client called Tycho, available for MacOS X, Linux and Windooze. You need an (apache) webserver with php 4.3 or better on the same box the voicemail is stored on. Before you can use the client you need to install the vmxml server scripts. The Stuff is beta but works pretty well. Were right now adding imap as transport layer so you wont need the server side php scripts in the future. The Stuff is available at: http://sip-syndication.com regards, Arnd ___ --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] Problem with Tycho Voicemail
Hello Sergio, please download and install the vmxml scripts again, there was a problem when php was configured with register_globals=off. This is fixed now. Please report success. best regards, Arnd ___ --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] Problem with Tycho Voicemail
Race Vanderdecken wrote: Is all the messages or just a message now and then? Sergio mailed me and said he cant listen to any voicemail. That was a stupid bug in our phpvoicemail script and not related to the infamous orphant .txt bug. There was a bug fix in 1.2.10 for orphaned .txt files in the /var/spool/asterisk/voicemail/.../INBOX directory. How old is your asterisk core? Were using 1.2.3 atm but we will upgrade to SVN with imap support soonish. Like I said just curious and might be way off base. But I am trying to track down the orphaned .txt file bug. As long as we still use 1.2.3. You want any infos send when we discover an orphant txt file? -- Arnd ___ --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] Problem with Tycho Voicemail
One more: be sure to set your webserver/php to register_globals off and safe_mode off, example from apaches httpd.conf: -- php_admin_flag register_globals off php_admin_flag safe_mode off -- Set this globally or in the virtual server config section. The scripts wont work with most installations when safe_mode is off best regards, Arnd --- Sergio R. D'Ippolito wrote: I’m using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Somebody use that software any time ? have you the same problem ? Thanks ___ --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] Problem with Tycho Voicemail
Arnd Vehling wrote: Set this globally or in the virtual server config section. The scripts wont work with most installations when safe_mode is off ^ wont work if safe_mode is ON! Damn, not my day today. ___ --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] Problem with Tycho Voicemail
Sergio R. D'Ippolito wrote: I’m using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Please send me or post here: - Client Version / os platform - Server Operating System - HTTP Server + php version - which version of the scripts you downloaded - your vmconfig.php config file without passwords! - if possible, look into your http server log for errors There is a known issue with sound not working on some Linux x86 platform not working reliably. ___ --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] Re: New Voicemail Client for Win32, Linux x86, MacOS X released
Steven wrote: Good point. If you click on that link twice by mistake, you may be deleting two different voicemails. As far as i tested it, its not possible because of the confirmation prompt coming up and the to-be-deleted message will be removed from the selection window before you can try to delete it again. Jordis comments about the renumbering problem are nevertheless correct and i will fix this bug as soon as i get my paid-contract work out of the way. I already had a look at the source of app_voicemail.c and must say that the voicemail storing system of asterisk seems to be, uhm, sub-optimal. -- Arnd ___ --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] New Voicemail Client for Win32, Linux x86, Mac OS X released
Hi, were releasing a beta version of our java eclipse based asterisk voicemail client called Tycho. You can download the client binary at: http://sip-syndication.com. Please note that the client needs a set of php scripts installed on the server side. The server side scripts, called vmxml are open source and available on the same website. Screenshots: MS Windows http://sip-syndication.com/images/stories/tycho/msw-tycho-overview-65.png Linux Mac OS X http://sip-syndication.com/images/stories/tycho/tycho_macosx_ppc_overview-65.png Linux http://sip-syndication.com/images/stories/tycho/lnx-ubuntu-tycho-overview-65.png If you have any questions or problems installing/using the client please mail: t y c h o at sip-syndication dot com or post to the support forum on our website. cheers, Arnd ___ --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] New Voicemail Client for Win32, Linux x86, Mac OS X released
Jordi Nelissen_ wrote: did not yet took the time to evaluate Tycho, but the Voicemail XML server side scripts are very helpfull. The source code is rather self-explanatory ... but it could be even more helpfull if you could provide us with some documentation on the usage of this API. The docs are somewhat basic but 90% complete. I will finish them start of next week. Check the website for an update of the vmxml docs. Ill post a followup in this thread when complete. cheers, Arnd ___ --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] Manager cmd: originate without picking up the fone?!
Hi There, we are developing a dialer application using the java lib to interface with the asterisk manager protocol. It works fine so far. The only problem we have is that if we use the originate command the user is required to pick up the fone _bevore_ asterisk will originate the call to the desired destination. What we would like to do is to place the call, check if the other end is available (ringing event) and only then let the user pickup the fone. Otherwise we would only display a text message Destination 'Busy' etc. Does anyone know if/how this is possible? cheers, Arnd ___ --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] Voicemail and musiconhold sound stopped working
Just FYI for anyone else who might run into this problem: After unloading the zaptel and zaprtc modules the audion works again! -- Arnd ___ 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] Voicemail and musiconhold sound stopped working
The no sound problem is very probably related to an incompatibility between the zaptel driver and the Teles AVM B1 ISDN card i am using. (kernel-capi 2.0) Anyone knows a work-around for this or is there no way to make both work? cheers, Arnd ___ 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] Voicemail and musiconhold sound stopped working
Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network sniffing using ngrep and verified that the voicemail app is indeed not sending _any_ udp/rtp packets towards my sip fones. I did restore old, working configs back but still no change. I reinstalled asterisk from the cvs and even rebootet my linux box (kernel 2.4.27) still no change. This stuff is now bugging me for 5 hours and i am slowly going nuts. I am using an installation with several different sip-fones, zaptel+zaprtc as well as fcpci+capi on a teles isdn card. Any ideas where to look for? thx, Arnd ___ 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] Comments on Areski Calling Card Solution plz
Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ___ 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] SER Config For Asterisk
Daniel Eboa wrote: what i want is be able to authenticate user before they connected to my asterisk box. users can be registered with asterisk, but i want that each time a user want to place outgoing call, he is first authenticate, and then authorize to place the call through the asterisk box. this is for billing meaning. This is a standard feature of SER. Just download the current SER stable release and configure (my)sql based authentication or radius authentication. After a successfull authentication just forward all calls to your asterisk box. This setup is pretty straight forward and u will find enough examples on the SER Homepage and the SER Mailing-List. There are even config templates for this included in the standard SER distribution. cheers, Arnd ___ 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] AS5300 + Asterisk
Hi, apenon apenon wrote: We have installed asterisk and using with many small ata. Now there is an AS5300 outside a PSTN PBX which makes termination. Now I want asterisk to handle only some of the calls with SIP and asterisk from 5300. There is no problem at asterisk configuration but in AS5300 how must I configure it for only some extension to register asterisk and forward call over it? dial-peer voice 123456 voip destination-pattern 3103464711 progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 2 session target ipv4:your.ip.address.goeshere This will route only one single number, u can set up several dial-peers with different patterns. cheers, Arnd ___ 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] Dial to a SIP fone ends up at Voicemail Busy
Hi, Terry H. Gilsenan wrote: I was having this problem with Gradstream BT101's with Asterisk @ Home version 0.7. The problem was that there was a sip channel still open (as far as asterisk and the phone were concerned) however this sip channel was not actually in use. The existence of this sip channel meant that whilst the phone could make calls, any incoming calls were directed to voicemail. Thanks for the hint. I did control the channels, they were all closed but the problem was still there. After testing the meeting app though (calling in via a PSTN-Cisco-Asterisk there is indeed a hung channel. Anyone knows what could be causing this? -- Channel (ContextExtensionPri ) State Appl. Data Zap/pseudo-1655835607 (defaults1 ) Rsrvd (None) (None) SIP/x.x.x.x-0814dbb8 (la-in 310 2 ) Up MeetMe |ip 2 active channel(s) -- cheers, Arnd ___ 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] Budgetone and NAT not working
Dan Morin wrote: Yes, I have both nat=yes and canreinvite=no. I have similiar setting (nat=route, canreinvite=no) and ive seen the same problems. My Server is on the internet though. I dont use any NAT support on the GS side and it does work most of the time. I havent seen this issue with my Sipura though. I noticed something very interesting today. Although it can not register, I can call from the budgetone to another extension. Same here. cheers, Arnd ___ 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] SER Config For Asterisk
Daniel Eboa wrote: This is the scenario i want to setup: Cisco ATA 186 --- SER - Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER should I use? I've try both ser-0.8.11 and ser-0.9.0 but seems like something is missing. I can't find some modules in ser.cfg file like: auth_radius.so and others. Can somebody help me with this issue?? Can u explain this in more detail please. Please make clear what u want 2 achieve. What u mean by and if there is a valid Account for the ATA? The ATA should be required to register and proxy-auth. If it can do this it must be either in the database or configured on a radius server. If it passes the auth u can just forward all calls to a asterisk box. cheers, Arnd ___ 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] Dial to a SIP fone ends up at Voicemail Busy
C F wrote: are these phones behind nat? Yes, but correctly registered. The same fones dont have any problems when registered to a SER Server. Can constantly reloading the configuration cause problems? cheers, Arnd ___ 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] Dial to a SIP fone ends up at Voicemail Busy
Hi, some of my sip fones which have several external numbers assigned are not reachable after a certain timespan. Instead of the fone the Voicemailbox is trigger in busy mode. After a reboot if the sip-fone the problem goes away for some time. Ive seen this problem with Sipuras and Grandstreams. Any ideas? cheers, Arnd ___ 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 + BudgeTone (behind NAT)
Hi, thats very probably a NAT problem. Your NAT box is probaly blocking the incoming UDP voice stream. If asteriks supports a RTP Proxy you can try that. best regards, Arnd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriks as SIPH323 Proxy?
Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC - SIP-Server - SIP/H323 Proxy - H323 Server - H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards, Arnd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
