[asterisk-users] SpiderMux?
Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Philip A. Prindeville wrote:
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,GotoIf($[${isnresult} != ]?:fn-CONGESTION,1)
; set up our outgoing call state
same = n,Set(SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} == ]?dial:)
same = n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
and the logging:
== ast_get_enum(num='555*9', tech='sip', suffix='freenum.org',
options='', record=1
== ENUM options(): pos=1, options='2'
== ISN ENUM: left=555, middle='9.'
== ast_get_enum() profiling: FAIL, 5.5.5.9.freenum.org, 21 ms
-- Executing [555*99...@outbound-freenum2:5] Set(SIP/guest_1-0010,
isnresult=) in new stack
-- Executing [555*99...@outbound-freenum2:6]
GotoIf(SIP/guest_1-0010, 0?:fn-CONGESTION,1) in new stack
-- Goto (outbound-freenum2,fn-CONGESTION,1)
[Apr 28 16:55:22] WARNING[5987]: pbx.c:4358 __ast_pbx_run: Channel
'SIP/guest_1-0010' sent into invalid extension 'fn-CONGESTION' in context
'outbound-freenum2', but no invalid handler
pbx*CLI
Note that the string fn-CONGESTION isn't matching the extension pattern:
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
and I'm not sure why.
Anyone want to venture how to go about figuring out how?
Hi
Try
exten = _fn-[A-Z].,1,NoOp(ISN: ${DIALSTATUS})
Ish
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[asterisk-users] Asterisk Query
Hi, I am new to asterisk and trying to make calls with TDM400P asterisk digium card. I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2 packages which are downloaded from asterisk website ( www.asterisk.org) and able to compile successfully. TDM400P Digium card (having only one FXS connected to J4) has installed successfully in PC. I would like to make calls across SIP [x-lite] to analog phone connected to TDM400P Digium card (fxs-j4). For this the following four conf files are modified as shown below. * chan_dahdi.conf* *==* [channels] context=test usecallerid=yes hidecallerid=no immediate=no signaling=fxo_ks echocancel=yes group=1 channel=1 *extensions.conf*** *=* [my-phones] exten = 2000,1,Dial(SIP/2000) [test] exten = ,1,Dial(Zap/1) exten = ,2,HangUp() *sip.conf*** *===* [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=my-phones secret=1234 host=dynamic *system.conf* *==* fxoks=1 loadzone = be defaultzone = be With those changes x-lite getting registered with asterisk and analog device/phone is getting ring tone with off-hook and also getting debug prints on cli, but not able to make calls. Test Setup: X-lite [configured as 2000, password… other info] running on asterisk PC àregistered with asterisk. Analog phone connected to TDM400P Digium card - FXS-J4 running on same asterisk PC à getting ring tone Test Result: = Tried by calling from x-lite à getting message on CLI “call from ‘2000’ to ‘’ rejected because extension not found” Tried by calling 2000 from analog phone [Digium-FXS-J4] - getting some engage/disconnected tone while pressing digts [2000] on phone itself. Welcome for your valuable suggestions and comments. Thank You in advance. Regards, Garge. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI == DeadAGI
Hello, i have this problem : i phone person B . *if i hang up*, i have this h extension : exten = h,1,AGI(ende.agi) *if the person B hangs up* , i have this h extension : exten = h,1, DeadAGI(ende.agi) The problem is, i do not know where hangs up the first . How kann i combine AGI and DeadAGI in one programm ? Thank you -- Mit freundlichen Gruessen. Redouane -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI == DeadAGI
Hello, i have this problem : i phone person B . *if i hang up,* i have this h extension : exten = h,1,AGI(ende.agi) *if the person B hangs up* , i have this h extension : exten = h,1, DeadAGI(ende.agi) The problem is, i do not know where hangs up the first . How kann i combine AGI and DeadAGI in one programm ? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect if a Number is up or not
Thanks Loan Indreias ... Nice Idea Thanks Danny Nicholas. Cheers On Tue, Apr 27, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote: This is probably a good idea, BUT it is likely that the dialed phone will never ring (Perhaps that is the desired effect); In my experience it takes Zap/DAHDI about 2-7 seconds to generate the first ring of a call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Tuesday, April 27, 2010 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Detect if a Number is up or not another idea you could test is to use a very short Timeout in your Dial command. like Dial(ZAP/012345678,1) - will dial and exit after 1 sec with DIALSTATUS set accordingly HTH, Ioan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI == DeadAGI
Redouane Zerargui wrote: Hello, i have this problem : i phone person B . _/*if i hang up*/_, i have this h extension : exten = h,1,AGI(ende.agi) _/*if the person B hangs up*/_ , i have this h extension : exten = h,1,DeadAGI(ende.agi) The problem is, i do not know where hangs up the first . How kann i combine AGI and DeadAGI in one programm ? Thank you -- Mit freundlichen Gruessen. Redouane Hi It is irrelevant who hangs up, you want to just use DeadAGI in the h extension Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mysql realtime schema
Hello. Where i can find complete realtime mysql schema for asterisk 1.6? Google get results to some tables. I want to do all iaxusers iaxpeers sipusers sippeers sipregs voicemail extensions meetme queues queue_members musiconhold queue_log in separate mysql tables. -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Starting call recording using a dynamic feature to call a macro
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when you listen later you can tell where the audio was paused.
So I changed things around so that instead of pausing and unpausing the
recording was stopped and then started again with a new file. An AGI
script would then join and mix the files together. The problem that I am
having is that I can stop the recording but when I ran the monitor
command again the recording didnt start.
So I decided to start with a simpler test and just have recording off by
default and then use #2 to start it but that doesnt work either. I
wondered if it was something to do with the native bridging so set
canreinvite=no on the test handsets I am using but no change.
Any ideas?
features.conf
[applicationmap]
pauseMonitor = #1,peer/callee,Macro,recpause,monitor-disabled
startMonitor = #2,peer/callee,Macro,recstart
unpauseMonitor = #3,peer/callee,Macro,recunpause,monitor-enabled
extensions.conf
[macro-recpause]
exten = s,1,Playback(disabled)
exten = s,n,PauseMonitor
[macro-recunpause]
exten = s,1,Playback(enabled)
exten = s,n,UnpauseMonitor
[macro-recstart]
exten = s,1,Set(FNAME=callrec_${MACRO_EXTEN}_${UNIQUEID}_GWTEST_${EPOCH})
exten = s,n,Monitor(wav,${FNAME},b)
[internal]
exten = 100,1,Dial(SIP/100,20)
exten = 110,1,Answer
exten =
110,n,Set(DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#startMonitor)
exten = 110,n,Set(FNAME=callrec_${EXTEN}_${UNIQUEID}_GWTEST_${EPOCH})
;exten = 110,n,Monitor(wav,${FNAME},b)
exten = 110,n,Dial(SIP/110,20)
exten = 110,n,Hangup
log :-
-- Executing [...@internal:1] Answer(SIP/100-0004, ) in new
stack
-- Executing [...@internal:2] Set(SIP/100-0004,
DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#testfeature#startMonitor)
in new stack
-- Executing [...@internal:3] Set(SIP/100-0004,
FNAME=callrec_110_1272534191.4_GWTEST_1272534191) in new stack
-- Executing [...@internal:4] Dial(SIP/100-0004,
SIP/110|20) in new stack
-- Called 110
-- SIP/110-0005 is ringing
-- SIP/110-0005 answered SIP/100-0004
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
-- Executing [...@macro-recstart:1] Set(SIP/100-0004,
FNAME=callrec_110_1272534191.4_GWTEST_1272534203) in new stack
-- Executing [...@macro-recstart:2] Monitor(SIP/100-0004,
wav|callrec_110_1272534191.4_GWTEST_1272534203|b) in new stack
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
== Spawn extension (internal, 110, 5) exited non-zero on
'SIP/100-0004'
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Re: [asterisk-users] Issue with (pattern) matching extension
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 29, 2010, at 1:33 AM, Ishfaq Malik wrote:
Philip A. Prindeville wrote:
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,GotoIf($[${isnresult} != ]?:fn-CONGESTION,1)
; set up our outgoing call state
same = n,Set(SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} == ]?dial:)
same = n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
and the logging:
== ast_get_enum(num='555*9', tech='sip', suffix='freenum.org',
options='', record=1
== ENUM options(): pos=1, options='2'
== ISN ENUM: left=555, middle='9.'
== ast_get_enum() profiling: FAIL, 5.5.5.9.freenum.org, 21 ms
-- Executing [555*99...@outbound-freenum2:5] Set(SIP/guest_1-0010,
isnresult=) in new stack
-- Executing [555*99...@outbound-freenum2:6]
GotoIf(SIP/guest_1-0010, 0?:fn-CONGESTION,1) in new stack
-- Goto (outbound-freenum2,fn-CONGESTION,1)
[Apr 28 16:55:22] WARNING[5987]: pbx.c:4358 __ast_pbx_run: Channel
'SIP/guest_1-0010' sent into invalid extension 'fn-CONGESTION' in
context 'outbound-freenum2', but no invalid handler
pbx*CLI
Note that the string fn-CONGESTION isn't matching the extension pattern:
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
and I'm not sure why.
Anyone want to venture how to go about figuring out how?
Hi
Try
exten = _fn-[A-Z].,1,NoOp(ISN: ${DIALSTATUS})
Ish
--
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Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Starting call recording using a dynamic feature to call a macro
Ignore me I figured it out. The dangers of copy and paste.
After looking through the code line by line I noticed the 'b' parameter
to monitor(). Fine to use before the dial command but shouldnt be used
when a call is in progress.
Gareth Blades wrote:
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when you listen later you can tell where the audio was paused.
So I changed things around so that instead of pausing and unpausing the
recording was stopped and then started again with a new file. An AGI
script would then join and mix the files together. The problem that I am
having is that I can stop the recording but when I ran the monitor
command again the recording didnt start.
So I decided to start with a simpler test and just have recording off by
default and then use #2 to start it but that doesnt work either. I
wondered if it was something to do with the native bridging so set
canreinvite=no on the test handsets I am using but no change.
Any ideas?
features.conf
[applicationmap]
pauseMonitor = #1,peer/callee,Macro,recpause,monitor-disabled
startMonitor = #2,peer/callee,Macro,recstart
unpauseMonitor = #3,peer/callee,Macro,recunpause,monitor-enabled
extensions.conf
[macro-recpause]
exten = s,1,Playback(disabled)
exten = s,n,PauseMonitor
[macro-recunpause]
exten = s,1,Playback(enabled)
exten = s,n,UnpauseMonitor
[macro-recstart]
exten = s,1,Set(FNAME=callrec_${MACRO_EXTEN}_${UNIQUEID}_GWTEST_${EPOCH})
exten = s,n,Monitor(wav,${FNAME},b)
[internal]
exten = 100,1,Dial(SIP/100,20)
exten = 110,1,Answer
exten =
110,n,Set(DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#startMonitor)
exten = 110,n,Set(FNAME=callrec_${EXTEN}_${UNIQUEID}_GWTEST_${EPOCH})
;exten = 110,n,Monitor(wav,${FNAME},b)
exten = 110,n,Dial(SIP/110,20)
exten = 110,n,Hangup
log :-
-- Executing [...@internal:1] Answer(SIP/100-0004, ) in new
stack
-- Executing [...@internal:2] Set(SIP/100-0004,
DYNAMIC_FEATURES=pauseMonitor#unpauseMonitor#testfeature#startMonitor)
in new stack
-- Executing [...@internal:3] Set(SIP/100-0004,
FNAME=callrec_110_1272534191.4_GWTEST_1272534191) in new stack
-- Executing [...@internal:4] Dial(SIP/100-0004,
SIP/110|20) in new stack
-- Called 110
-- SIP/110-0005 is ringing
-- SIP/110-0005 answered SIP/100-0004
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
-- Executing [...@macro-recstart:1] Set(SIP/100-0004,
FNAME=callrec_110_1272534191.4_GWTEST_1272534203) in new stack
-- Executing [...@macro-recstart:2] Monitor(SIP/100-0004,
wav|callrec_110_1272534191.4_GWTEST_1272534203|b) in new stack
-- Packet2Packet bridging SIP/100-0004 and SIP/110-0005
== Spawn extension (internal, 110, 5) exited non-zero on
'SIP/100-0004'
--
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[asterisk-users] Polycom 330 not connecting
I was just wondering if anyone is having the same problem will Polycom 330 ip phone. The phone looses the network and when you reboot the phone it can no longer find the DHCP server. I put an address in manually, but the phone is still not able to connect to the network. I replaced the phone with another phone (with the same settings) everything works. The computer attached to the phone still works even through the phone does not. There have been no changes in the environment. The phones (3 in 1 day) are only 1.5 years old. I have tried setting the phone back to factory default, but still the same problem. Any ideas? Tony LaMear Systems Support Specialist Indianapolis Zoo www.indianapoliszoo.comhttp://www.indianapoliszoo.com/ The Indianapolis Zoo empowers people and communities, both locally and globally, to advance animal conservation. [cid:image001.png@01CAE772.402EFE30] Before printing this e-mail, think green and conserve paper inline: image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No change in payload. (SDP)
Aditya Kumar wrote: re-posting the question. --- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload. I want it to send as it is to the external proxy. How can I achieve this? so that the SDP/payload will not be modified for users talking to the external world. I want media for those external devices to come Directly to the users in my pbx. (with out going t asterisk) 2) also related question is can I have the xml payload in the originator and call is routed via PBX to the Target. The xml payload also must be carried to the target. is it possible This will really help me as I was held up with this :( Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as such the outgoing INVITE is a *different* session from the incoming one. That means that Asterisk has to be able to understand the SDP content that arrives so it can forward media between the two sessions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming call should ring on several dahdi channels
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I change it to
exten = 12345678,1,Dial(DAHDI/13DAHDI/14)
DAHDI/13 and 14 rings together, but I can't add the ,,rtT features (the
biggest problem is the person who picked up the call can't transfer to
another extension)
If anybody have an idea how to solve this problem, please let me know.
Thanks for you help in advance.
Best regards,
Peter Gelencser
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Re: [asterisk-users] B400P and A1200P changes card order
2010.04.20. 16:50 keltezéssel, Shaun Ruffell írta: On 04/19/2010 03:48 AM, Peter Gelencser wrote: I've run into a veird problem. I'm using a B400P BRI and an A1200P card with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and spans, everything seems fine. With dahdi_genconf I made the config, set up the channels in chan_dahdi.conf, but after reboot, the two card changes the order, (originaly the B400P is the first (closer to the processor) and A1200P is the second), when I check with dahdi_scan, it shows the channels are not in the original order. I set the Card ID Switch to 0 on B400P but it does not solve the problem. I tried these cards with another motherboard, the situation is the same so it's not chipset specific. Please let me know if there is any solution. Thank you for your help in advance. Peter, I'm not familiar with the A1200P, but I suspect the A1200P not blacklisted in /etc/modprobe.d. Therefore when you reboot, it's driver is loaded by udev before the rest of the DAHDI drivers are loaded. Try adding the driver for the A1200P to /etc/modprobe.d/dahdi.blacklist to make sure that the drivers are loaded by any configuration scripts in /etc/init.d in a consistent order. Cheers, That was the solutions, thanks for the tip. Best regards, Peter Gelencser -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call should ring on several dahdi channels
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I change it to
exten = 12345678,1,Dial(DAHDI/13DAHDI/14)
DAHDI/13 and 14 rings together, but I can't add the ,,rtT features (the
biggest problem is the person who picked up the call can't transfer to
another extension)
If anybody have an idea how to solve this problem, please let me know.
Thanks for you help in advance.
Best regards,
Peter Gelencser
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Re: [asterisk-users] incoming call should ring on several dahdi channels
typo ...
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}${OFFICE2},,rtT)
Gareth Blades wrote:
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I change it to
exten = 12345678,1,Dial(DAHDI/13DAHDI/14)
DAHDI/13 and 14 rings together, but I can't add the ,,rtT features (the
biggest problem is the person who picked up the call can't transfer to
another extension)
If anybody have an idea how to solve this problem, please let me know.
Thanks for you help in advance.
Best regards,
Peter Gelencser
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Re: [asterisk-users] Issue with (pattern) matching extension
Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
Right! Be very careful about what you're matching! When it comes to matching
things like 'N', 'X', 'Z', etc... the case does not matter!
exten = _f[n]-.,1,NoOp(...)
That'll fix you up.
Leif.
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Re: [asterisk-users] Issue with (pattern) matching extension
Good snippet, Leif. It's easier to read 100 threads on this forum than the
100 pages of the infamous Asterisk Book PDF.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, April 29, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issue with (pattern) matching extension
Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
Right! Be very careful about what you're matching! When it comes to matching
things like 'N', 'X', 'Z', etc... the case does not matter!
exten = _f[n]-.,1,NoOp(...)
That'll fix you up.
Leif.
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Re: [asterisk-users] Dropping incompatible voice frame
Possibly or possibly not. Most (IMO) calls are placed initially with the choice 2-3 or more codecs. Normally one codec is negotiated and life goes on, but IAX is a little different from a SIP/DAHDI call. The most certain remedy I can think of for this it to just unallow the alaw codec on IAX calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, April 29, 2010 8:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dropping incompatible voice frame Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping incompatible voice frame
Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Invite issue
Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308 _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Invite issue
Can you post a sip debug Tarek Sawah wrote: Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308 _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
Also, the prefix _stdexten-. seems to work fine in the [stdexten]
context, so I'm not sure what's different here.
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Re: [asterisk-users] Issue with (pattern) matching extension
It's a pattern matching thing; the asterisk module knows how to process
stdexten, but thinks that n or N is a digit substitution. When n or
N is escaped ([n] or [N]) the program knows to treat it as a literal and
not a pattern match.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philip
Prindeville
Sent: Thursday, April 29, 2010 11:47 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Issue with (pattern) matching extension
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
Also, the prefix _stdexten-. seems to work fine in the [stdexten]
context, so I'm not sure what's different here.
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Re: [asterisk-users] Odd Issue With Polycom Phones]
The phone is only making one call, notice the call-id did not change. The second INVITE is sent in responce to a 401 Authentication Required. The 401 will contain the necessary authentication information for the phone to use to build the Authorization header that it inserts in the second invite. THe mechanism uses a shared secret (the reg.X.auth.userId and reg.X.auth.password in the polycom cfg file, and the secret=X and the userID(I think thats what its called) in the asterisk config files). If you have other phones that are not doing this second invite I would bet its because on the asterisk side you have not configured them to use a secret. -- Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3...@y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD From: 3271 sip:3271@ y.y.y.y sip:3...@y.y.y.y;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone CSeq: 1 INVITE Call-ID: 96a1fe9c-88f06c73-7e209...@x.x.x.x Contact: sip:3271@ x.x.x.x:5060 sip:3...@x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3261@ y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008 From: 3271 sip:3271@ y.y.y.y sip:3...@y.y.y.y;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone CSeq: 2 INVITE Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x Contact: sip:3271@ x.x.x.x:5060 sip:3...@x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3261@ y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phone sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. Thanks. -Jay -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Tuesday, April 20, 2010 4:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Odd Issue With Polycom Phones -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote: On Wed, 28 Apr 2010, Ryan Bullock wrote: Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early in the script to read in anything from stdin? (From the docs) # pull AGI variables into %input %input = $AGI-ReadParse(); early == before (any interaction with Asterisk || exit) Any reason Asterisk::AGI shouldn't do that automatically? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
On Wed, 28 Apr 2010, Ryan Bullock wrote: Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early in the script to read in anything from stdin? (From the docs) # pull AGI variables into %input %input = $AGI-ReadParse(); On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote: early == before (any interaction with Asterisk || exit) On Thu, 29 Apr 2010, Tzafrir Cohen wrote: Any reason Asterisk::AGI shouldn't do that automatically? A good question for the author of Asterisk::AGIwrap. When I wrote my AGI library for C, I thought about it but ultimately didn't. If I ever get around to publishing my code I would add it just to raise the success rate for first time users. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
Speaking from a Perl'er perspective, there's no good reason that Asterisk::AGI shouldn't do the ReadParse automatically except that it requires the module author to do something that the user should be doing as a best practice and could lead to unexpected errors in a reuse environment. IMO there could be more and better Perl modules out there for use, but I think that most serious Asterisk users probably take Steve's advice and leave Perl for C once they pass point X. That or they are masochistic PHP users :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, April 29, 2010 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script On Wed, 28 Apr 2010, Ryan Bullock wrote: Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early in the script to read in anything from stdin? (From the docs) # pull AGI variables into %input %input = $AGI-ReadParse(); On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote: early == before (any interaction with Asterisk || exit) On Thu, 29 Apr 2010, Tzafrir Cohen wrote: Any reason Asterisk::AGI shouldn't do that automatically? A good question for the author of Asterisk::AGIwrap. When I wrote my AGI library for C, I thought about it but ultimately didn't. If I ever get around to publishing my code I would add it just to raise the success rate for first time users. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote:
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
No, it's both cases.
Also, the prefix _stdexten-. seems to work fine in the [stdexten]
context, so I'm not sure what's different here.
Actually, it probably doesn't. Testing fail.
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Re: [asterisk-users] Issue with (pattern) matching extension
On 04/29/2010 12:09 PM, Tilghman Lesher wrote:
On Thursday 29 April 2010 11:46:39 Philip Prindeville wrote:
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
No, it's both cases.
Also, the prefix _stdexten-. seems to work fine in the [stdexten]
context, so I'm not sure what's different here.
Actually, it probably doesn't. Testing fail.
That's highly inconsistent, then.
Why does the 'n' in stdexten (or the 'x' for that matter) match a
regular letter in that case, but not in the freenum case?
Doesn't quite make it 'deterministic' if you have to test it to see what
it's going to do.
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Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI script
Un-top-posting... On Wed, 28 Apr 2010, Ryan Bullock wrote: Looking at the Asterisk::AGI docs, maybe try calling ReadParse() early in the script to read in anything from stdin? (From the docs) # pull AGI variables into %input %input = $AGI-ReadParse(); On Wed, Apr 28, 2010 at 09:34:18AM -0700, Steve Edwards wrote: early == before (any interaction with Asterisk || exit) On Thu, 29 Apr 2010, Tzafrir Cohen wrote: Any reason Asterisk::AGI shouldn't do that automatically? A good question for the author of Asterisk::AGIwrap. When I wrote my AGI library for C, I thought about it but ultimately didn't. If I ever get around to publishing my code I would add it just to raise the success rate for first time users. On Thu, 29 Apr 2010, Danny Nicholas wrote: Speaking from a Perl'er perspective, there's no good reason that Asterisk::AGI shouldn't do the ReadParse automatically except that it requires the module author to do something that the user should be doing as a best practice and could lead to unexpected errors in a reuse environment. IMO there could be more and better Perl modules out there for use, but I think that most serious Asterisk users probably take Steve's advice and leave Perl for C once they pass point X. That or they are masochistic PHP users :) The OP's code was using Asterisk::AGIwrap. I'm not a Perl weenie, but I think this is an in-house package -- Google can't find anything relevant. Adding a simple have I been initialized? check at the start of each function would help the first time user and would not lead to a reuse issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1) in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail(IAX2/pbx2-15464,
u8029) in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame*
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
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Re: [asterisk-users] Code in extensions.conf to leave a voice mail inanother PBX ?!
If you dial 8029 from PBX1, does VM work? In my experience, cross-version
IAX is tricky.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Code in extensions.conf to leave a voice mail
inanother PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1) in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail(IAX2/pbx2-15464,
u8029) in new stack
[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
_
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Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!
In PBX1, where are you actually dialing the phone? The first line of the
macro just says goto dialstatus with no Dial statement.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Code in extensions.conf to leave a voice mail in
another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1) in
new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1] VoiceMail(IAX2/pbx2-15464,
u8029) in new stack
[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed
to write frame
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on 'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
_
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Re: [asterisk-users] Asterisk Query
2010/4/29 garge rama garge.r...@gmail.com
Hi,
I am new to asterisk and trying to make calls with TDM400P asterisk digium
card.
I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
libpri-1.4.10.2 packages which are downloaded from asterisk website (
www.asterisk.org)
and able to compile successfully. TDM400P Digium card (having only one FXS
connected to J4) has installed successfully in PC.
I would like to make calls across SIP [x-lite] to analog phone connected to
TDM400P Digium card (fxs-j4).
For this the following four conf files are modified as shown below.
* chan_dahdi.conf*
*==*
[channels]
context=test
usecallerid=yes
hidecallerid=no
immediate=no
signaling=fxo_ks
echocancel=yes
group=1
channel=1
*extensions.conf***
*=*
[my-phones] ---*EXTEN does not exists for your sip
peer context*
exten = 2000,1,Dial(SIP/2000)
; Should look like:
*exten = ,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you
want})
[test]
exten = ,1,Dial(Zap/1)
exten = ,2,HangUp()
*sip.conf***
*===*
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
*context=**my-phones *
secret=1234
host=dynamic
*system.conf*
*==*
fxoks=1
loadzone = be
defaultzone = be
With those changes x-lite getting registered with asterisk and analog
device/phone is getting ring tone with off-hook and also getting debug
prints on cli, but not able to make calls.
Test Setup:
X-lite [configured as 2000, password… other info] running on asterisk PC
à registered with asterisk.
Analog phone connected to TDM400P Digium card - FXS-J4 running on same
asterisk PC à getting ring tone
Test Result:
=
Tried by calling from x-lite à getting message on CLI “call from
‘2000’ to ‘’ rejected because extension not found”
Tried by calling 2000 from analog phone [Digium-FXS-J4] - getting some
engage/disconnected tone while pressing digts [2000] on phone itself.
Welcome for your valuable suggestions and comments. Thank You in advance.
Regards,
Garge.
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--
Juan.
Linux User #441131
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[asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:
[macro-stdexten]
exten = s,n,Dial(SIP/${ARG1}IAX2/${ar...@${arg1},20,tTrWw);Ring phone
for 20 seconds
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
2010/4/29 Peder pe...@networkoblivion.com
In PBX1, where are you actually dialing the phone? The first line of the
macro just says “goto dialstatus” with no Dial statement.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati
*Sent:* Thursday, April 29, 2010 2:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten = 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i have:
exten = 8029,1,Macro(stdexten,8029)
and in stdexten macro:
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
-- Executing [...@macro-stdexten:6] Goto(IAX2/pbx2-15464, s-NOANSWER|1)
in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-noans...@macro-stdexten:1]
VoiceMail(IAX2/pbx2-15464, u8029) in new stack
*[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
Failed to write frame*
-- IAX2/pbx2-15464 Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'
== Spawn extension (default, 8029, 1) exited non-zero on
'IAX2/pbx2-15464'
-- Hungup 'IAX2/pbx2-15464'
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
the issue I'm having, thanks a lot!
--
Abdullah
--
_
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--
Abdullah
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Re: [asterisk-users] Issue with (pattern) matching extension
On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: Doesn't quite make it 'deterministic' if you have to test it to see what it's going to do. The code is deterministic. The human who wrote the example is not. Are you proposing a genetic modification to make humans deterministic? -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Worse things have been proposed for humans; many readers would like to see this done to posters such as I. :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, April 29, 2010 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issue with (pattern) matching extension On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: Doesn't quite make it 'deterministic' if you have to test it to see what it's going to do. The code is deterministic. The human who wrote the example is not. Are you proposing a genetic modification to make humans deterministic? -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On 4/29/10 1:55 PM, Tilghman Lesher wrote: On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: Doesn't quite make it 'deterministic' if you have to test it to see what it's going to do. The code is deterministic. The human who wrote the example is not. Are you proposing a genetic modification to make humans deterministic? If we're going to examine gene therapy, let's start with suppressing the polemic gene, shall we? Rather than your committing a point fix to stdexten which was reported as a side-effect, you might also have looked over my proposed fix which covered both, reviewed that, and committed that instead. Oh, well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On Thu, 29 Apr 2010, David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... PAP2T - excellent Handytone - crap Pretty much every large scale TSP has standardized on the PAP2T or 2102. There is a reason the Handytone is priced so low... j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On Thu, 29 Apr 2010, David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've been wondering about the pap2t's. However, im struggling to get the provisioning documentation. Still waiting for Computer2000 (techdata) to send it over.. 3 months waiting. However, avoid the handytones! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On 4/29/2010 3:10 PM, Jeff LaCoursiere wrote: On Thu, 29 Apr 2010, David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... PAP2T - excellent Handytone - crap Pretty much every large scale TSP has standardized on the PAP2T or 2102. There is a reason the Handytone is priced so low... j I agree with Jeff. I use a PAP2T for home (one SIP connection to Broadvoice and the other to our work Asterisk) and has worked near flawlessly. My first ATA was the HT286, and just does not hold a candle. Dean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
Dan Journo wrote: On Thu, 29 Apr 2010, David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... I've been wondering about the pap2t's. However, im struggling to get the provisioning documentation. Still waiting for Computer2000 (techdata) to send it over.. 3 months waiting. However, avoid the handytones! Dan You can get the guide on the INTERNET! http://track.sipfoundry.org/secure/attachment/16445/SPA_Provisioning_v3.pdf :) -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
On Thursday, April 29, 2010, David Backeberg wrote: What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. I can't claim the bonus points. However, I did have a couple of Grandstream 286s until one failed and I bought a PAP2T to replace both. Best buy I made in a long time as it was very easy to set up via the web interface and it's proved rock solid reliable. In contrast, the Grandstream 286s had to be rebooted about once a week and often suffered from audio dropouts. HTH, -- Geoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls Dropping
Hi, I'm having a major problem with random calls dropping. After spending weeks trying to figure it out, i've finally spotted the issue but don't know how to resolve it. I run a sip server that's hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat router. Every so often, all the rtp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i'm not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk stopping for no reason
Hi, Few days ago, my asterisk began to stop unexpectedly What I did: - Added a mp3 to the musiconhold directory - Adjusted the permissions (chown asterisk:asterisk + chmod 755) - Reconfigured the musiconhold.conf to the deprecated format (found the example on the internet) [classes] default = quietmp3:/etc/asterisk/moh,r - Restarted the service I thought the new mp3 was corrupted so I removed it from the server. The problem perssisted so yesterday I changed the deprecated configuration to: [default] mode=quietmp3 directory=/etc/asterisk/moh random=yes My original configuration was: [default] mode=files directory=/etc/asterisk/moh I have no logs telling me thate quietmp3 failed and I cannot find any way to see if the musiconhold was enabled when asterisk dropped. Here are my questions (finally): Do you know if quietmp3 may kill the server? Is there a way to set random in files mode? I am using Asterisk 1.4.17~dfsg-2ubuntu1.1 on an Ubuntu 8.04.4 server. Thank you very much --- Alexandre Vézina -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On 04/29/2010 03:56 PM, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) Leif. Danny: Well, there is an effort to improve the documentation. See the asterisk-doc mailing list. If there are particular issues you'd like to see resolved, take them up there. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me to my answering machine :-(
Bryan Jacobs wrote: I can't just call the car - the car is my cell phone DID with a bluetooth kit. I did this same thing you're attempting. I have a desk set at home, a Polycom in my office and my cell phone all being called at the same time. I called Verizon and had them disable voice mail on my cell phone so that the only voice mail system I use is my Asterisk box. I no longer give out my cell phone number but only my home phone number and allow Asterisk to do all of the heavy lifting. Oh, and I set the caller*ID outbound to the caller*ID of the inbound call so I can still see who it is. Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Not really complaining, but AKAIK, this document is current as of about 1.4.10? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philip A. Prindeville Sent: Thursday, April 29, 2010 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with (pattern) matching extension On 04/29/2010 03:56 PM, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) Leif. Danny: Well, there is an effort to improve the documentation. See the asterisk-doc mailing list. If there are particular issues you'd like to see resolved, take them up there. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On 29 Apr 2010, at 22:56, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) He's insulting our holy book! Stone him! ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Certainly getting people to email in updated examples would speed the book along... On 04/29/2010 04:06 PM, Danny Nicholas wrote: Not really complaining, but AKAIK, this document is current as of about 1.4.10? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philip A. Prindeville Sent: Thursday, April 29, 2010 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with (pattern) matching extension On 04/29/2010 03:56 PM, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) Leif. Danny: Well, there is an effort to improve the documentation. See the asterisk-doc mailing list. If there are particular issues you'd like to see resolved, take them up there. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
Are you guys talking about the Asterisk Cookbook Because that could be released in the next 20 years at this point... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Apr 29, 2010 at 6:14 PM, Steve Howes steve-li...@geekinter.net wrote: On 29 Apr 2010, at 22:56, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) He's insulting our holy book! Stone him! ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me to my answering machine :-(
On Thu, 29 Apr 2010 18:03:26 -0400 Barry L. Kline blkl...@attglobal.net wrote: Bryan Jacobs wrote: I can't just call the car - the car is my cell phone DID with a bluetooth kit. I did this same thing you're attempting. I have a desk set at home, a Polycom in my office and my cell phone all being called at the same time. I called Verizon and had them disable voice mail on my cell phone so that the only voice mail system I use is my Asterisk box. I no longer give out my cell phone number but only my home phone number and allow Asterisk to do all of the heavy lifting. That's a great idea. I think I'd port my cell phone number to the Asterisk box and get a new cell number assigned which nobody knows. This is the best solution I've heard so far - no hacks at all! I wonder if all the cell providers let you do this? Oh, and I set the caller*ID outbound to the caller*ID of the inbound call so I can still see who it is. I already did this with the 'o' dial option. Barry Bryan Jacobs signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Duplicated DTMF with bridged IAX channels maybe?
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX * PSTN ---* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated every so often. I've attached an IAX trace from the PSTN server to this message where you can see the duplication (digits 9 3). The digits entered were 258963. [snip] Testing the following scenario: Call --(from pstn PRI)-- (PSTN box) --IAX-- (PBX box) --IAX-- (PSTN box) --(to/from pstn PRI)-- (PSTN box) Results in Duplication. Here are 2 traces from the PSTN box's DTMF log: '8' was not duplicated, '9' was. Zap/42-1 is the first inbound leg from the PSTN IAX2/w2bpstn-8399 is the leg from the PBX box to the PSTN box Zap/53-1 is the inbound leg from the PSTN [Apr 29 17:02:15] DTMF[16062] channel.c: DTMF begin '8' received on Zap/42-1 [Apr 29 17:02:15] DTMF[16062] channel.c: DTMF begin passthrough '8' on Zap/42-1 [Apr 29 17:02:15] DTMF[16065] channel.c: DTMF begin '8' received on IAX2/w2bpstn-8399 [Apr 29 17:02:15] DTMF[16065] channel.c: DTMF begin passthrough '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:15] DTMF[16071] channel.c: DTMF begin '8' received on Zap/53-1 [Apr 29 17:02:15] DTMF[16071] channel.c: DTMF begin ignored '8' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '8' received on Zap/42-1, duration 63 ms [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end accepted with begin '8' on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '8' has duration 63 but want minimum 80, emulating on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end emulation of '8' queued on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end '8' received on IAX2/w2bpstn-8399, duration 0 ms [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end accepted with begin '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end passthrough '8' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end '8' received on Zap/53-1, duration 223 ms [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end passthrough '8' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF begin '9' received on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF begin passthrough '9' on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF begin '9' received on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF begin passthrough '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin '9' received on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin ignored '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end '9' received on Zap/53-1, duration 223 ms [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF end passthrough '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin '9' received on Zap/53-1 [Apr 29 17:02:16] DTMF[16071] channel.c: DTMF begin ignored '9' on Zap/53-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '9' received on Zap/42-1, duration 63 ms [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end accepted with begin '9' on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end '9' has duration 63 but want minimum 80, emulating on Zap/42-1 [Apr 29 17:02:16] DTMF[16062] channel.c: DTMF end emulation of '9' queued on Zap/42-1 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end '9' received on IAX2/w2bpstn-8399, duration 0 ms [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end accepted with begin '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:16] DTMF[16065] channel.c: DTMF end passthrough '9' on IAX2/w2bpstn-8399 [Apr 29 17:02:17] DTMF[16071] channel.c: DTMF end '9' received on Zap/53-1, duration 223 ms [Apr 29 17:02:17] DTMF[16071] channel.c: DTMF end passthrough '9' on Zap/53-1 -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Continuing after a TIMEOUT(absolute)
Greetings,
I'm trying to continue to do some processing after a TIMEOUT
(absolute). In my dialplan below, when a call comes in to [default],
I call macro-phonenum and pass it a timeout of 20 seconds. macro-
phonenum sets TIMEOUT(absolute), then loops saying the phone number
that was called (in MACRO_EXTEN). When the timeout expires I want to
call my macro-hangup (so it can say goodbye or whatever). But the
system is just hanging up. The dialplan and log output is below. Any
info is appreciated. This is on version 1.6.0.5.
[macro-answer-and-join]
exten = s,1,NoOp()
exten = s,n,Answer()
exten = s,n,Wait(4)
exten = s,n,SendDTMF(1)
exten = s,n,Wait(1)
exten = s,n,SendDTMF(1)
exten = s,n,MacroExit
[macro-hangup]
exten = s,1,NoOp()
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
;
exten = T,1,NoOp()
exten = T,n,Playback(goodbye)
exten = T,n,Hangup()
[macro-phonenum]
exten = s,1,NoOp()
exten = s,n,Macro(answer-and-join)
exten = s,n,Set(TIMEOUT(absolute)=${ARG1})
exten = s,n,Set(i=1000)
exten = s,n,While($[${i} = 1])
exten = s,n,SayDigits(${MACRO_EXTEN})
exten = s,n,Wait(5)
exten = s,n,Set(i=$[${i} - 1])
exten = s,n,EndWhile()
exten = s,n,MacroExit
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
exten = T,n,MacroExit
[default]
exten = _X.,1,NoOp()
exten = _X.,n,Macro(phonenum,20)
exten = _X.,n,Macro(hangup)
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
The log when the timeout occurs:
snip (I'm in macro-phonenum)
-- SIP/70.124.61.17-082a69a8 Playing 'digits/5.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- Executing [...@macro-phonenum:7] Wait(SIP/
70.124.61.17-082a69a8, 5) in new stack
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8' in macro 'phonenum'
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8'
Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK)
set_destination: Parsing sip:70.124.61.17:5060 for address/port to
send to
set_destination: set destination to 70.124.61.17, port 5060
Reliably Transmitting (NAT) to 70.124.61.17:5060:
BYE sip:70.124.61.17:5060 SIP/2.0
snip
Cheers,
- Brendan
Brendan Sterne
QA Lead, Callvine
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[asterisk-users] Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 iPhone
Hi, If I was going to post this as an iPhone-only SIP client, I'd expect loud booing and hissing, but Media5 mobile SIP client is available for the Symbian S60 platform, too, or will be shortly. Interested? To join us and hear about Media5 form Pascal Dore, see http://vuc.me Speaking of mobile, if any of you know the people who are developing the SIP client for Android, we'd love to hear from them ASAP. Our SMS/Voicemail line is +1 (518) VUC-VOIP = +1 518 882 8647. If you were reading this in Chrome with the phone number extension, that phone number would be clickable. Here's that extension: Google Voice - Google Chrome extension gallery http://vuc.li/dj0zuv You can call sip:200...@login.zipdx.com in g722 wideband (or g711) Jump on IRC #vuc channel on Freenode.net There are a number of other ways to connect including an iNum and more SIP URI. Check the site for details on the header of the home page. For the exact time in your time zone: http://vuc.me/next Hope to hear you there. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID on Asterisk and Astribank
Hi all...
I have a problem with caller id on my asterisk server.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)
everything fine until I try to feed my app with caller id.
My extensions.conf :
[incoming1]
exten = s,1,AGI(/var/apps/core/runagi,incoming,${CALLERID(num)})
exten = s,n,QUEUE(${que},trkd)
exten = h,1,Hangup()
here is the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit
found in fsk data.
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed
failed: Success
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID
returned with error on channel 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-- Playing 'en/0006' (escape_digits=) (sample_offset 0)
I read instructions from a few forums
then I made a change on 'chan_dahdi.conf' like :
-
1:
cidsignalling=v23, cidstart=ring, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18
(Ring Begin)...
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 2
(Ring/Answered)...
[Apr 30 11:42:05] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18
(Ring Begin)...
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:42:06] WARNING[31296]: chan_dahdi.c:6174 dahdi_handle_event:
Ring/Off-hook in strange state 6 on channel 15
== Spawn extension (default, s, 2) exited non-zero on 'DAHDI/15-1'
-
2:
cidsignalling=dtmf', cidstart=ring, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:28] WARNING[31491]: chan_dahdi.c:8610 ss_thread: DTMFCID timed
out waiting for ring. Exiting simple switch
-- Hungup 'DAHDI/15-1'
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:34] DEBUG[31492]: chan_dahdi.c:8630 ss_thread: CID is '', flags 8
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:49:35] WARNING[31492]: chan_dahdi.c:6174 dahdi_handle_event:
Ring/Off-hook in strange state 6 on channel 15
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-
3:
cidsignalling=dtmf', cidstart=polarity, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Registered IAX2 '9009' (AUTHENTICATED) at 127.0.0.1:48961
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-
4:
cidsignalling=v23', cidstart=polarity, hidecallerid=no,
callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1]
