Re: [asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
On Tue, 7 Apr 2009, George Pajari wrote: I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has no PCI boards). *8 Call Pickup works fine from any of the phones connected using the Linksys SPA2102. *8 Call Pickup does not work from the Cisco 7940G phones (chan_sip.c:13977 handle_request_invite: Nothing to pick up for 000d6556-eeb3001c-76b88543-7f51d...@192.168.0.211) Seems someone else had the same problem back in 2004 and got no answer. http://lists.digium.com/pipermail/asterisk-users/2004-April/036869.html Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk is not designed for University with largeuser base?
On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is not designed for University with large user base?
Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] oslec + dahdi
On Thu, 22 Jan 2009, troxlinux wrote: I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn I have installed oslec and loaded, but it doesn't work me with dahdi modinfo oslec filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 13813ACD4A228F69FF4B5C1 depends: vermagic: 2.6.18-92.1.22.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4. oslec is a great great great software, with the version of zaptel 1.4.11 I had it installed and without anything of echo in my card TDM 400 I almost have the same enviroment as you, I basically run the following script to get oslec work with my tdm411 card. #!/bin/sh cd /usr/src wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 tar xjf linux-2.6.28.tar.bz2 wget http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-2.1.0.2.tar.gz wget http://downloads.digium.com/pub/telephony/dahdi-linux/dahdi-linux-2.1.0.3.tar.gz tar zxvf dahdi-linux-2.1.0.3.tar.gz ln -s /usr/src/dahdi-linux-2.1.0.3 /usr/src/dahdi mkdir /usr/src/dahdi/drivers/staging cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild cd /usr/src/dahdi make make install cd /usr/src tar zxvf dahdi-tools-2.1.0.2.tar.gz cd /usr/src/dahdi-tools-2.1.0.2 ./configure make make install Hope it helps. Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hearing transfer during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
[ht286]
type=friend
regexten=6010
username=ht286
secret=secret
context=numberplan-local
callerid=Home Phone 6010
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
[EMAIL PROTECTED]
dtmfmode=rfc2833
extensions.conf:
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[default]
exten = s,1,Ringing
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(if-u-know-ext-dial)
exten = s,n,Background(otherwise)
exten = s,n,Background(to-reach-operator)
exten = s,n,Background(pls-hold-while-try)
exten = s,n,WaitExten(6)
exten = s,n,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
include = internal
[internal]
; define local extensions here
exten = 6010,1,Macro(stdexten,${EXTEN},SIP/ht286)
[numberplan-local]
ignorepat = 9
include = default
include = parkedcalls
comment = Local Calling
include = internal
features.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
;context = numberplan-local; Which context parked calls are in
;parkingtime = 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout = 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot = next ; Continue to the 'next' parking
space. Defaults to 'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = # ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record (a.k.a. Touch Monitor)
atxfer = * ; A
users.conf:
[6004]
fullname = Analog User 4
secret = 6004
email =
cid_number =
zapchan = 4
context = numberplan-local
hasvoicemail = yes
hasdirectory = yes
hassip = no
hasiax = no
hasmanager = no
callwaiting = no
threewaycalling = no
mailbox = 6004
hasagent = no
group = 2
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[asterisk-users] Hearing transfer during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
[ht286]
type=friend
regexten=6010
username=ht286
secret=secret
context=numberplan-local
callerid=Home Phone 6010
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
[EMAIL PROTECTED]
dtmfmode=rfc2833
extensions.conf:
[macro-stdexten]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[default]
exten = s,1,Ringing
exten = s,n,Wait(1)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,Background(if-u-know-ext-dial)
exten = s,n,Background(otherwise)
exten = s,n,Background(to-reach-operator)
exten = s,n,Background(pls-hold-while-try)
exten = s,n,WaitExten(6)
exten = s,n,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,n,Hangup()
include = internal
[internal]
; define local extensions here
exten = 6010,1,Macro(stdexten,${EXTEN},SIP/ht286)
[numberplan-local]
ignorepat = 9
include = default
include = parkedcalls
comment = Local Calling
include = internal
features.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
;context = numberplan-local; Which context parked calls are in
;parkingtime = 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout = 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot = next ; Continue to the 'next' parking
space. Defaults to 'first' available
;pickupexten = *8 ; Configure the pickup extension. Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = # ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record (a.k.a. Touch Monitor)
atxfer = * ; A
users.conf:
[6004]
fullname = Analog User 4
secret = 6004
email =
cid_number =
zapchan = 4
context = numberplan-local
hasvoicemail = yes
hasdirectory = yes
hassip = no
hasiax = no
hasmanager = no
callwaiting = no
threewaycalling = no
mailbox = 6004
hasagent = no
group = 2
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[asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line
Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it achievble in Asterisk to let them make PSTN call simulataneously through one sinle POTS line? I don't know anything about traditional PBX system, it seems one shop can have one single phone number and mutiple extensions, then the extensionss can make/receive PSTN call simultaneously, is this the same senerio as the one single POTS line to FXO and multiple extensions on FXSs? Thanks for help. Vincent ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A simple IVR extension problem
Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf context=incoming signalling=fxs_ks channel = 4 context=internal signalling=fxo_ks channel = 1 - extensions.conf: [office] exten = s,1,Dial(Zap/1,30) [home] exten = s,1,Macro(stdexten,106,SIP/ht286,t) [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,1,Background(enter-ext-of-person) exten = s,n,WaitExten(20) exten = 100,1,Dial(Zap/1,30) exten = 106,1,Macro(stdexten,106,SIP/ht286) exten = 101,1,Macro(stdexten,101,SIP/vli) exten = 107,1,AGI(math.agi) exten = 108,1,Playback(12) ;exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) ;exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) When I call my PSTN number, I can hear the enter-ext-of-person message, but once I press any one of the extension number, Asterisk sometime execute the relevant extension application, sometime not at all. If I comment the IVR extensions config and simply use : exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) I can always get call My console message: ( Asterisk did not execute relevant extension in the last two call after I entered the extension digit) -- Starting simple switch on 'Zap/4-1' [Aug 2 13:46:38] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:46:40] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Macro(Zap/4-1, stdexten|101|SIP/vli|t) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, SIP/vli|20) in new stack -- Called vli -- SIP/vli-08353298 is ringing -- SIP/vli-08353298 answered Zap/4-1 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:47:32] NOTICE[4437]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:47:33] ERROR[4437]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-168) [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, Zap/1|30) in new stack -- Called 1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/4-1 -- Native bridging Zap/4-1 and Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (incoming, 100, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:48:22] NOTICE[]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:48:23] ERROR[]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-9) [Aug 2 13:48:23] WARNING[]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:48:23] WARNING[]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] AGI(Zap/4-1, math.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/math.agi -- Playing 'math-game-welcome' (escape_digits=) (sample_offset 0) -- Playing 'math-game-next' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/17' (language 'en') -- Playing 'add' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/15' (language 'en') -- Zap/4-1 Playing 'equals' (language 'en') -- Playing 'math-game-wrong' (escape_digits=) (sample_offset 0) -- Playing 'math-game-your-answer' (escape_digits=) (sample_offset 0) -- Zap/4-1 Playing 'digits/0' (language 'en') -- Playing 'math-game-right-answer' (escape_digits=) (sample_offset 0)
