Re: [Asterisk-Users] My Boss wants background music!!!!
Why not just dial an extention for music when the user wants music from there desk. >The requirement of the original poster was to mute the music at the desk >when a call is in progress. > I>t would be really nice if there was a hardphone capable of accepting a >multicast high-quality stream when no call was in progress. ___ 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
Re: [Asterisk-Users] How can i test a modem with Asterisk?
Question, What makes them useless? What does the xp100 have that others are missing, just wondering. thanks >Clue: most modems, even voice ones, are useless for something like >Asterisk. The 537 modems are supported largely because they can be. ___ 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
Re: [Asterisk-Users] Cannot Start Asterisk
did u issue this command ? "modprobe zaptel" before the asterisk -vvvgc if u got fxo modprobe wcfxo as well On Mon, 29 Nov 2004 18:50:53 -0800, Norman Zhang <[EMAIL PROTECTED]> wrote: > Hi, > > I'm running asterisk-1.0.2-2mdk. When I tried to start it with > /usr/sbin/asterisk -gc, I get > > [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) > Ouch ... error while writing audio data: : Broken pipe > > # ps aux | grep mpg123 > root 5237 0.1 0.4 5816 pts/0S18:45 0:00 mpg123 -q > -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 > fpm-world-mix.mp3 > > Could someone please give me a few pointers on how to troubleshoot this? > > /var/log/asterisk/messages > > Unable to get our IP address, Skinny disabled > Unable to open pseudo channel for timing... > Sound may be choppy. > Unable to open IAX timing interface: No such file or directory > Unable to get our IP address, Skinny disabled > > Regards, > Norman Zhang > ___ > 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 > -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.ca ___ 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
Re: [Asterisk-Users] IP to IP call without server?
I agree you can do this with SIP. but I would use skype, msn, yahoo or VOIP blasters (get on ebay) for a simple call to call without a server. its too much effort and too much to learn for a simple call. On Mon, 29 Nov 2004 04:07:43 +0900, nkb <[EMAIL PROTECTED]> wrote: > Hi. > I'm really new. > I was just wondering if it is possible at all to do a IP to IP call > without a * server (or as a matter of fact, any other kind of server)? > say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at > hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must we > all both be registered with the same server to do that? Can this not be > done without passing thru server (*)? > Thanks. > ___ > 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 > -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.ca ___ 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
Re: [Asterisk-Users] asterisk & xlite codecs
hi there, How about changing the general conf in sip. > disallow=all > allow=ulaw > allow=alaw > allow=gsm not just disallow=all and take them out of the extentions conf. To me since you have the same codecs allowed its kinda not needed in my mind to specify it to that level. Maybe it will fix your problem 2? On Thu, 11 Nov 2004 12:47:07 -, Ashling O'Driscoll <[EMAIL PROTECTED]> wrote: > Hello, > > I am having problems getting two xlite clients to communicate through > asterisk. I am getting an error message: > > chan_sip.c:2753 process_sdp: No compatible codecs. > > I have enabled all possible codecs in xlite (Menu -> Advanced system > settings -Codec settings) and have added the appropriate lines in > sip.conf (see below) to allow all codecs. However this is still not > working. I have looked this problem up on google and it was > previously attributed to old versions of asterisk. However I dont > have asterisk setup long and got the most recent version of it. > > Please help if possible. I must get a call working soon, > Kindest Regards, > Aisling. > > sip.conf > > [general > > port=5060 > bindaddr=0.0.0.0 > disallow=all > > ;xlite client one > > [2000] > type=friend > username=2000 > secret=bla > regexten=2000 > nat=yes > auth=md5 > context=from-sip > callerid="Aisling"<2000> > dmtfmode=rfc2833 > canreinvite=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=dynamic > mailbox=2000 > > ;xlite client two > > [2001] > type=friend > username=2001 > secret=bla2 > regexten=2001 > nat=yes > auth=md5 > context=from-sip > callerid="Julien"<2001> > dmtfmode=rfc2833 > canreinvite=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > host=dynamic > mailbox=2001 > > ---Legal Disclaimer--- > > The above electronic mail transmission is confidential and intended only for > the person to whom it is addressed. Its contents may be protected by legal > and/or professional privilege. Should it be received by you in error please > contact the sender at the above quoted email address. Any unauthorised form > of reproduction of this message is strictly prohibited. The Institute does > not guarantee the security of any information electronically transmitted and > is not liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the sender nor > for any delay in its receipt. > > > > ___ > 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 > -- Regards, Steven Kalcevich Office +1- 416-576-4457 MSN: [EMAIL PROTECTED] http://www.ciscokid.net http://www.sohonetworks.ca ___ 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
