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Asterisk Questions - beginners Moderator: Moderators Users browsing this forum: None		Goto page 1, 2, 3 ... 45, 46, 47 Next
	Forum Index -> Asterisk Questions - beginners	Mark all topics read

Topics		Replies	Author	Views	Last Post
	Announcement: Rules	0	Arcopix	3226	Fri May 18, 2007 4:59 pm Arcopix
	Sticky: TrixBox 2.2 with Idefisk/Zoiper 2.0	6	Hellark	4521	Mon May 04, 2009 10:37 pm AAVoIP.com
	Problem starting asterisk	0	kwcummings	12	Thu Mar 11, 2010 1:45 pm kwcummings
	DAHDI Channel bank issues	0	kc7gr	5	Thu Mar 11, 2010 6:12 am kc7gr
	WIll Pay: Need help in setting up asterisk Server & SIP	0	aligak	37	Tue Mar 09, 2010 7:04 pm aligak
	Issue with Asterisk n-way call	1	srireddy	15	Tue Mar 09, 2010 11:10 am srireddy
	UUIE content	0	Werz	6	Mon Mar 08, 2010 12:27 pm Werz
	2 ASTERISK BOXES TRUNKED BEHIND NAT (INTERNET)	0	asiawatcher	35	Sat Mar 06, 2010 12:31 am asiawatcher
	Stop mixmonitor in my channel	0	leonardo07746	9	Fri Mar 05, 2010 3:43 pm leonardo07746
	Trunk sip	0	kloklo	16	Thu Mar 04, 2010 11:41 am kloklo
	Atcom IP-BRIM - Poor Quality + No Call Pickup + No CallerID	1	pmcorp	39	Wed Mar 03, 2010 12:32 pm pmcorp
	Asterisk CDR Report problem	0	marco	18	Mon Mar 01, 2010 10:45 am marco
	Calls disconnect once put on hold	0	ob5000	35	Sun Feb 28, 2010 4:06 am ob5000
	Problems in Grandstream Asterisk SRTP connection	0	hemanshurpatel	39	Sat Feb 27, 2010 7:08 am hemanshurpatel
	Problems in Grandstream Asterisk SRTP connection	0	hemanshurpatel	15	Sat Feb 27, 2010 7:04 am hemanshurpatel
	Identify a Phone's mac address?	0	rommelraj	26	Fri Feb 26, 2010 3:07 pm rommelraj
	Problems with trunk sip asterisk	0	tivoli6	25	Fri Feb 26, 2010 2:55 pm tivoli6
	sending IP address	0	davidexe	30	Wed Feb 24, 2010 10:22 pm davidexe
	extensions.conf not loading	0	elvis.nuno	24	Wed Feb 24, 2010 5:39 pm elvis.nuno
	Manager initiated (Back to Back) Call Scenario	0	moeness86	38	Mon Feb 22, 2010 12:51 pm moeness86
	How to cancel an auto dial-out in progress?	1	machtnix	622	Fri Feb 19, 2010 4:13 pm jacco van Tuijl
	RTP point to point	0	nokitec	35	Fri Feb 19, 2010 10:48 am nokitec
	how to use T to transfer call by called party	0	dxb.pawan	43	Thu Feb 18, 2010 9:04 pm dxb.pawan
	Gsm Gateway (MV-370) cannot always call through	0	nerdje25	54	Thu Feb 18, 2010 3:42 pm nerdje25
	Post to PHP on call connect	0	josephrbf	47	Thu Feb 18, 2010 2:57 pm josephrbf
	Sip Trunk with Telesinergy	0	ecfenix	31	Thu Feb 18, 2010 2:32 pm ecfenix
	Cannot Playback MP3 - Asterisk 1.6.2	0	nandinibhaduri	33	Thu Feb 18, 2010 12:59 pm nandinibhaduri
	Cannot Playback MP3 - Asterisk 1.6.2	0	nandinibhaduri	34	Thu Feb 18, 2010 12:57 pm nandinibhaduri
	call parking	0	dxb.pawan	45	Wed Feb 17, 2010 10:23 am dxb.pawan
	Picking and holding calls through software	0	nacef	54	Tue Feb 16, 2010 6:07 pm nacef
	How to handle broadband line dropouts	0	proware	63	Tue Feb 16, 2010 7:23 am proware
	Playback start position	0	arpitrmodi	44	Mon Feb 15, 2010 11:12 am arpitrmodi
	Meetme and sound quality	0	m.victor	47	Sat Feb 13, 2010 12:57 pm m.victor
	Generate two calls, 1 to play recording other to record	0	mattux	52	Fri Feb 12, 2010 10:07 am mattux
	Incomming Call ISsues	0	psabilly	46	Thu Feb 11, 2010 10:11 pm psabilly
	Asterisk SIP server	0	thesti	89	Thu Feb 11, 2010 1:49 am thesti
	Question about connecting channel bank to asterisk	0	pyouwsy	77	Tue Feb 09, 2010 11:33 pm pyouwsy
	Identify CAPI Channel on a BRI interface	0	kilogrammo	64	Fri Feb 05, 2010 7:30 pm kilogrammo
	How do i add user extensions other than 6000	1	kameelperdza	69	Fri Feb 05, 2010 4:17 pm sadzas
	Voice Guard Control Protocol- Driver for VoIP Anti- Blocking	0	speed-voip.com	54	Fri Feb 05, 2010 8:15 am speed-voip.com
	Voice Guard Control Protocol- Driver for VoIP Anti- Blocking	0	speed-voip.com	41	Fri Feb 05, 2010 8:15 am speed-voip.com
	include with time-condition does not work	0	guildo	55	Wed Feb 03, 2010 11:52 pm guildo
	Can't hear remote extension Ringing when i call	0	deadhuman	61	Wed Feb 03, 2010 10:49 pm deadhuman
	Video support: SIP to H323	0	GPM	65	Wed Feb 03, 2010 12:49 pm GPM
	Video support: SIP to H323	0	GPM	48	Wed Feb 03, 2010 12:48 pm GPM
	Video support: SIP to H323	0	GPM	44	Wed Feb 03, 2010 12:48 pm GPM
	Caller ID passthrough issue	1	neftv	130	Mon Feb 01, 2010 8:32 pm neftv
	Asterisk Manager - Call picked up from Hold	0	ellisgl	56	Mon Feb 01, 2010 3:11 am ellisgl
	Unable to create channel of type 'SIP'	0	normaldotcom	54	Sat Jan 30, 2010 3:42 am normaldotcom
	Which .conf file do I use?	0	aureolevoyager	61	Fri Jan 29, 2010 8:07 pm aureolevoyager
	Trunk / Context problem	1	sopel1000	127	Fri Jan 29, 2010 8:05 pm sopel1000
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