Steven J. Sobol
2003-Aug-26 09:36 UTC
[Asterisk-Users] More questions. Call Waiting and Threeway
I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking something (plus, I'm still not up to speed on the hardware/telco end of the setup - all of the work I'm doing is with software). Is there any way to control whether three-way and caller ID are enabled per-call or per-SIP-phone? What I'd like to do is, for example, be able to dial *70 from my SIP phone to turn off call waiting, or be able to enable three-way on a per-phone basis. I don't know what's on the other side of the Zap channels (i.e. PRI, CT-1, whatever), but if it makes a difference, I can find out. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.net
Hey all, I'm current experiencing a problem when attempting to make a speex call from X-Lite to Asterisk... When I attempt to call an extension with the demo configuration, I get no audio (only with speex), and after awhile my /var/log/asterisk/messages fills up with this: Aug 26 12:18:53 WARNING[122894]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Here is a brief cut from the messages log for the entire transaction.. Aug 26 12:22:08 DEBUG[57352]: File chan_sip.c, Line 3783 (check_user): Setting NAT on RTP to 0 Aug 26 12:22:08 DEBUG[57352]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '094423C8-03B7-48AD-A557-C389F0AD126E@198.186.202.182' of Response 30335: Found Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 3783 (check_user): Setting NAT on RTP to 0 Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 4807 (handle_request): Check for res Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 952 (find_user): Call from user 'nettwerk' is 1 out of 0 Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop: <sip:nettwerk@198.186.202.182:5060> Aug 26 12:22:09 DEBUG[139278]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from UNKN to SPEEX Aug 26 12:22:09 DEBUG[57352]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '094423C8-03B7-48AD-A557-C389F0AD126E@198.186.202.182' of Response 30336: Found Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space Aug 26 12:22:23 WARNING[139278]: File codec_speex.c, Line 167 (speextolin_framein): Out of buffer space I *know* speex is supported by asterisk, so I must be doing something dumb.. lol Any help would be appreciated -san