RE: [Asterisk-Users] VAD/DTX implementation through zaptel cards
Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. mail2web - Check your email from the web at http://mail2web.com/ . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot open chan_zap:
Hello, On Sun, 10 Apr 2005, Tim Connolly wrote: I'm working on getting a new Digium TE110XP working. I no_load the chan_zap module, otherwise * doesn't start. When I try to load it manually I see: pbx01*CLI load chan_zap Unable to load module chan_zap Apr 10 22:13:23 WARNING[4349]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap: cannot open shared object file: No such file or directory Yet, * running as root, the file exists: -rwxr-xr-x 1 root root 296495 Apr 10 17:33 /usr/lib/asterisk/modules/chan_zap.so* Any suggestions? Run ldd /usr/lib/asterisk/modules/chan_zap.so maybe some library chan_zap.so wants to load is missing. Regards Torsten -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 4 x ISDN2 hardware...?
Hi, I've done some testing with asterisk and I must say I'm very impressed by all the features. Now I want to create a production environment and am looking into all the available ISDN cards. The cards I've found are: 1. AVM C4 (1300 euro's) 2. Eicon Diva with 4 ISDN2 ports (even more expensive) 3. Junghanss card with 4 ISDN2 ports (600 euro's) Besides the voice part, I would also like to be able to receive and send faxes. Which card is best? If I understand it correctly, the junghanss card is a 4 port HFC card. I tested Asterisk with another 1 port HFC card and rxfax, but found out that not all faxes are received correctly. As my business is depending on faxes, I find it very important that the incoming faxes are received correctly. Does anybody have experience with receiving faxes with the AVM C4? I assume that it is fully supported by Asterisk and the analog modem to receive faxes as implemented in the hardware of the AVM C4? Any information would be appreciated. Thanks! Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 x ISDN2 hardware...?
It's my opinion that whilst asterisk indeed has some fax capability, it's not a business-grade fax platform. If faxes are indeed as important to your business as you suggest, I'd be inclinded to look for alternatives. - Original Message - From: Marc [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 4:29 PM Subject: [Asterisk-Users] 4 x ISDN2 hardware...? Hi, I've done some testing with asterisk and I must say I'm very impressed by all the features. Now I want to create a production environment and am looking into all the available ISDN cards. The cards I've found are: 1. AVM C4 (1300 euro's) 2. Eicon Diva with 4 ISDN2 ports (even more expensive) 3. Junghanss card with 4 ISDN2 ports (600 euro's) Besides the voice part, I would also like to be able to receive and send faxes. Which card is best? If I understand it correctly, the junghanss card is a 4 port HFC card. I tested Asterisk with another 1 port HFC card and rxfax, but found out that not all faxes are received correctly. As my business is depending on faxes, I find it very important that the incoming faxes are received correctly. Does anybody have experience with receiving faxes with the AVM C4? I assume that it is fully supported by Asterisk and the analog modem to receive faxes as implemented in the hardware of the AVM C4? Any information would be appreciated. Thanks! Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callback application
I don't know if what you're trying to do is possible, but the easiest way to check would be to take a look at the raw packets on the ethernet interface of your * server once a call is in progress. If indeed the RTP can be handed off to the 2 endpoints, you should only see SIP traffic at your server. TCPDUMP is your friend. - Original Message - From: snacktime [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 1:32 PM Subject: [Asterisk-Users] Callback application I wasn't sure how else to label this thread because I'm not sure on the correct terminology to use when decribing what I'm trying to do... I am using livevoip and have a DID with them also, both using SIP. THe big picture is that I'm making a callback application. Right now I'm testing out a couple of things just using DISA. What I'm trying to do is setup a two legged call using * and DISA, with both legs going to/from livevoip, and set the call up in a way where the voice traffic goes straight between livevoip/livevoip once both legs are established. What I don't know is how to tell if I have succeeded in this. Using the following I get both legs up and * say's it's created a native bridge between the two legs. However a 'sip show channels' still shows both channels in *. How do I tell if the voice data is not going through * anymore? Basically once the legs are joined, with one originating from livevoip and one terminating to livevoip, I want my * box out of the picture as far as the voice data stream goes. [outgoing] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) [from-livevoip] exten = 800xxx,1,Ringing exten = 800xxx,2,Wait(1) exten = 800xxx,3,Answer exten = 800xxx,4,DISA(no-password|outgoing) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callback application
On Mon, 2005-04-11 at 16:40 +1000, Rod Bacon wrote: I don't know if what you're trying to do is possible, but the easiest way to check would be to take a look at the raw packets on the ethernet interface of your * server once a call is in progress. If indeed the RTP can be handed off to the 2 endpoints, you should only see SIP traffic at your server. TCPDUMP is your friend. or sip debug, or iptraf/jnettop/any other network traffic monitor. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 x ISDN2 hardware...?
Is it possible to use hylafax and asterisk with only the AVM C4? Or do I need a separete fax modem? -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Rod Bacon Verzonden: maandag 11 april 2005 8:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] 4 x ISDN2 hardware...? It's my opinion that whilst asterisk indeed has some fax capability, it's not a business-grade fax platform. If faxes are indeed as important to your business as you suggest, I'd be inclinded to look for alternatives. - Original Message - From: Marc [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 4:29 PM Subject: [Asterisk-Users] 4 x ISDN2 hardware...? Hi, I've done some testing with asterisk and I must say I'm very impressed by all the features. Now I want to create a production environment and am looking into all the available ISDN cards. The cards I've found are: 1. AVM C4 (1300 euro's) 2. Eicon Diva with 4 ISDN2 ports (even more expensive) 3. Junghanss card with 4 ISDN2 ports (600 euro's) Besides the voice part, I would also like to be able to receive and send faxes. Which card is best? If I understand it correctly, the junghanss card is a 4 port HFC card. I tested Asterisk with another 1 port HFC card and rxfax, but found out that not all faxes are received correctly. As my business is depending on faxes, I find it very important that the incoming faxes are received correctly. Does anybody have experience with receiving faxes with the AVM C4? I assume that it is fully supported by Asterisk and the analog modem to receive faxes as implemented in the hardware of the AVM C4? Any information would be appreciated. Thanks! Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setgroup Checkgroup
I have some troubles to use Setgroup / Checkgroup!!!
I setup a test (NoOP's are deleted): First caller should get first line,
second caller should get second line, third caller should get busy and
send an email. Note, that I used twice here to check the first line!!!
[trunkint_A]
exten = _90N.,104,SetGroup(sip-13); increase Group counter
exten = _90N.,105,CheckGroup(1); check no more than 1
in this group
exten = _90N.,106,NoOp(Line 106)
exten = _90N.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,108,hangup
;
exten = _90N.,206,SetGroup(sip-12)
exten = _90N.,207,CheckGroup(1)
exten = _90N.,208,NoOp(Line 208)
exten = _90N.,209,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,210,hangup
;
exten = _90N.,308,SetGroup(sip-13)
exten = _90N.,309,CheckGroup(1)
exten = _90N.,310,NoOp(Line 310)
exten = _90N.,311,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _90N.,312,hangup
;
exten = _90N.,410,Busy
exten = _90N.,411,SYSTEM(mail -s 'VPBX all lines in use'
[EMAIL PROTECTED])
I thought that 104 will set the Group counter sip-13 to 1 and will
use line 107 for the dial command
If another caller comes in that way, sip-13 would be 2 and because
Checkgroup allows only 1, the Group coutner would be setback to 1
and it will follow the jump to 206 and sets the Group counter sip-12
to 1
A third call should now find Group counter sip-12 and sip-13 set to 1
and give a busy signal and send an email.
HOWEVER, the log file show:
-- Executing SetGroup(Local/[EMAIL PROTECTED],2, sip-13)
in new stack
-- Executing CheckGroup(Local/[EMAIL PROTECTED],2, 1) in
new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Line 106)
in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
so far so good!
-- Executing SetGroup(SIP/615-92c3, sip-13) in new stack
-- Executing CheckGroup(SIP/615-92c3, 1) in new stack
-- Executing NoOp(SIP/615-92c3, Line 106) in new stack
-- Executing Dial(SIP/615-92c3, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
Ahh, it does not check Group counter sip-13, ... it checks SIP/615-92c3
and Local/[EMAIL PROTECTED],2
How can I make it that it checks exactly the Group countersip-13
bye
Ronald
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[Asterisk-Users] TDM400P power supply
Hello All! I've a problem with a TDM400P digium card. My box has no molex connectors for power supply. Simply has no any power connector, because is not a normal PC) And I need to know if i can use a external supply. But I've several questions: 1.- Are both circuits (PCI-power and Phone-line-power) electrically separated? 2.- A little voltage difference can create an undesired internal current? 3.- What are the current needs for this supply? I need the power supply because I want to use both FXS and FXO ports. And I can't use a Y cable, because I've no molex connectors. Thanks in advance. Rpr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 x ISDN2 hardware...?
Marc wrote: Is it possible to use hylafax and asterisk with only the AVM C4? Or do I need a separete fax modem? Works fine and dandy with a single AVM C4 here. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat changing the sound of dialtone
On Sun, Apr 10, 2005 at 09:47:36AM -0500, Andy Hamilton wrote: On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote: Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. This depends on what kind of phone you are using. Sorry - With standard POTS phones on a Digium TDM FXS interface. Thanks, Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN with Asterisk
Hi 1) Don't bother considering analogue lines. Too problematic and not any cheaper in the long term. 2) the HFC chipset ISDN cards at £13 are fine as long as you make sure you assign each card its own IRQ in the bios. http://www.komplett.co.uk/k/ki.asp?sku=119006cks=SPK I have 3 of these cards running in a Celeron 450. 2 cards in NT mode for ISDN phones and 1 connected to my BT ISDN 2e line. You will need bristuffed asterisk from www.junghanns.com to drive the cards. They use native zaptel drivers. DDI and caller ID on ISDN are all supported natively in Asterisk. Works perfectly in PTP and PTMP modes. Best regards Tim Robinson Basingstoke, UK Henry Owens wrote: Hi all, I'm currently tasked with implementing a low-cost, high performance and reliable telephone system for a motorcycle dealership in the UK, and Asterisk is my primary candidate for the system. My question is: can Asterisk work well as a small office (8 extensions) PBX, with a mixture of analogue and IP phones, on an ISDN2e telephone line from BT? The reason i am thinking of using ISDN2e is that i need to be able to have two lines, but with one number (i.e. if someone calls on the main number and stays on the call, and then a second person calls the same number, instead of an engaged tone, they will get through and another staff member can take the call). If there is another way to do this with anaologue lines, i'm open to suggestions. I have looked at using a service such as sipgate.co.uk to allow this to happen, however the fact that i can not list the number in a normal telephone directory would be prohibitive (though i will probably want to connect to a VoIP service for outbound calls also). If it does work well with ISDN, will features such as DDI (direct dial inwards) and caller id be possible? And can anyone recommend any hardware that will allow me to connect my asterisk PBX to the ISDN line? Many thanks for any information you can offer! Regards, Henry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferance DialPlan
I'd like to make a dial plan but couldn't work it out. I'd be appreciated if
you can help me.
The client reaches asterisk by PRI and starts conferance by the SIP agent
dedicated to his number. besides, I want to add another second client who
dialed the same number to the first client's conferance by the SIP agent.
the point is this: I call from PRI with SIP agent by the dial but they start
the conferance without entering the conferance room. when 2 call come enters
the conferance room being aware of that the SIP is busy. I need to meet the
calls and SIP in the same conferance room.
Here is my current Conferance Dial Plan
[conferance]
exten = _XX,1,Ringing(10)
exten = _XX,2,Answer
exten = _XX,3,SetGlobalVar(numara=${EXTEN})
exten = _XX,4,Dial(SIP/${EXTEN},30,m)
exten = _XX,5,Goto(${numara}-${DIALSTATUS},1)
exten = _XX,6,Meetme(${numara})
exten = _XX-BUSY,1,Meetme(${numara})
exten = _XX-ANSWER,1,Meetme(${numara})
exten = _XX-NOANSWER,1,Playback(jingle)
exten = _XX-NOANSWER,2,Hangup
exten = _XX-CHANUNAVAIL,1,Playback(jingle)
exten = _XX-CHANUNAVAIL,2,hangup
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Re: [Asterisk-Users] 4 x ISDN2 hardware...?
On Monday 11 April 2005 08:29, Peer Oliver Schmidt wrote: Marc wrote: Is it possible to use hylafax and asterisk with only the AVM C4? Or do I need a separete fax modem? Works fine and dandy with a single AVM C4 here. Just wanted to chip in to say that Eicon's Diva Server 4BRI-8M is working great in a combined CAPI + TTY mode ... Asterisk listens with CAPI, HylaFAX uses the TTY interface.. Of course, we only have a need to send faxes so that simplifies the setup :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PTSN POTS Differences SOLVED
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten = _1NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _1NXXNXX,2,Macro(outisbusy); No available circuits
exten = _9.,1,Macro(dialout-trunk,1,w${EXTEN:1})
exten = _9.,2,Macro(outisbusy); No available circuits
exten = _NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXXNXX,2,Macro(outisbusy); No available circuits
exten = _NXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXX,2,Macro(outisbusy); No available circuits
Couldn't you have just put the w in once, in the Dial command that
is inside [macro-dialout-trunk] ?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] IAX calls between asterisk boxes works 1 way only
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'
Error, so I obviously missed something and can someone smack me upside
the head and point out my error.
Please assume that the passwords are correct in the files :-).
Configurations are attached of each box:
Box 1
iax.conf
[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=default
callerid=Guest IAX User
[salisbury]
type=friend
host=192.168.254.100
username=northbuild
secret=password
context=voip
permit=192.168.254.100
extensions.conf
[global]
PSTNLine=Zap/g1
AnalogPhone=Zap/g2
[pstn]
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,4,VoiceMail(u690)
exten = s,5,Hangup
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
exten = 1690,1,VoicemailMain,s690
exten = 1691,1,VoicemailMain,s691
[outgoing]
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup
[sip]
exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr)
exten = 690,3,Voicemail2,u690
exten = 690,103,Voicemail2,b690
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr)
exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691
include = internal
include = outgoing
include = parkedcalls
include = voip
[voip]
exten = _1XX,1,Dial(IAX2/james:password@192.168.254.100/${EXTEN})
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
-
Box 2
iax.conf
[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=default
callerid=Guest IAX User
[dixst]
type=friend
host=192.168.50.1
username=dixst
secret=password
context=e100p
permit=192.168.50.1
[james]
type=friend
host=192.168.69.1
username=james
secret=password
context=e100p
permit=192.168.69.1
extensions.conf
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[e100p]
exten = _1XX,1,Dial(Zap/g1/${EXTEN})
exten = _93X.,1,Dial(Zap/g1/${EXTEN})
exten = _9073X.,1,Dial(Zap/g1/${EXTEN})
include = dialstring
include = voip
[voip]
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
; DixSt Redcliffe Ext
exten = _66X,1,Dial(IAX2/scarb:password@192.168.60.1/${EXTEN})
; Scarborough
exten = _69X,1,Dial(IAX2/northbuild:password@192.168.69.1/${EXTEN})
; James Home
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Re: [Asterisk-Users] iax / realtime problems
Hi Mat, Did the following: 1. Upgraded to new CVS HEAD version CVS-NHEAD-04/11/05-16:08:03 On the Makefile, enabled the ff: # Optional debugging parameters DEBUG_THREADS = -DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS MALLOC_DEBUG = -include $(PWD)/include/asterisk/astmm.h I cannot seem to enable pg on this line in Makefile #Include debug symbols in the executables (-g) and profiling info (-pg) DEBUG=-g #-pg I get error below when I do make valgrind gcc: -pg and -fomit-frame-pointer are incompatible I skip enabling pg and continue with make clean and make valgrind. gdb backtrace still gives vague output: (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0048087a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00304420 in ?? () #5 0x0100 in ?? () #6 0x0049f900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00304580 in ?? () #8 0x00599605 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x0049f8fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00304840 in ?? () #11 0x0048074c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so Still not clear, any pointers to make the backtrace more verbose? On Mon, 2005-04-11 at 00:05, Matthew Boehm wrote: In order for this to be helpful, you need to recompile with make valgrind and edit your Makefile and turn on all the debugging stuff. -Matthew From: Paul P. Pongco [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 9 Apr 2005 15:13:55 +0800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] iax / realtime problems Hi Mat, I can easily replicate the problem. I just put an entry on the iax table for mysql, fire up iax soft client and BOOM .. asterisk core dumps. What's weird is sip is working fine using realtime. Here is a gdb backtrace. Not really a programmer. Hope someone helps. Thanks. #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 (gdb) bt #0 0x00beeec0 in vfprintf () from /lib/tls/libc.so.6 #1 0x00c0f286 in vsnprintf () from /lib/tls/libc.so.6 #2 0x00bf7622 in snprintf () from /lib/tls/libc.so.6 #3 0x0031187a in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #4 0x00b19340 in ?? () #5 0x0100 in ?? () #6 0x00330900 in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #7 0x00b19480 in ?? () #8 0x0082d5d6 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #9 0x003308fc in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #10 0x00b19720 in ?? () #11 0x0031174c in ?? () from /usr/lib/asterisk/modules/res_config_mysql.so #12 0x in ?? () On Apr 8, 2005 8:53 PM, Matt Schulte [EMAIL PROTECTED] wrote: I've never actually core dumped but I *have* been able to hang asterisk a couple times, I believed my problem was when I lost my mysql connection. Why it lost connection is a mystery, the servers are on the same testswitch. :/ I forgot which head ver it was, a couple weeks ago. -Original Message- From: Paul P. Pongco [mailto:[EMAIL PROTECTED] Sent: Friday, April 08, 2005 1:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax / realtime problems Hello, I am using CVS-NHEAD-03/29/05-15:51:16 and testing iax realtime. I have configured a test account on iax.conf: [test] type=friend context=test username=test auth=md5 secret=testing host=dynamic disallow=all allow=ilbc allow=gsm callerid=1010 trunk=no qualify=no Then I insert an entry on mysql for testing realtime (btw realtime on the asterisk box works well for sip on both the flatfile and mysql). It has the same config as that on the flatfile but with different username and password (iaxtest). Asterisk crashes with the following error: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03403 DCall: 0 [x.x.0.93:4569] USERNAME: iaxtest REFRESH : 300 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 3 DCall: 03403 [x.x.0.93:4569] -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 --snip, above lines just repeat here-- -- Seeding 'iaxtest' at x.x.0.93:4569 for 60 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) On iax.conf rtcachefriends=yes rtnoupdate=yes rtautoclear=yes
[Asterisk-Users] Snom 'virtual' extension monitoring?
Hi list! I'm working to replace a PBX with group ring indication. On the current PBX each phone has 3 buttons with a light to identify an incoming call ringing for a certain group. For example if the phone is ringing at sales a led lights up to indicate a call coming in on that group (but that phone is not ringing). Other departments can hear the phone ringing in the other dept. even though it's not their own phone that is ringing. Can I use the programmable buttons on a Snom phone to achieve a similar thing? I was thinking of creating a virtual extension for each group (like sales, admin, support etc). For each department I include that extension to 'ring'. The snom would monitor this extension for incoming unanswered called and people in another dept. can answer the call by just picking up the phone. Is this possible? Can a virtual extension be created this way and can it be monitored in the way I was thinking of? This would be an alternative to the groups feature of *, because I don't think you can use the buttons and lights on the snom phones in the way I described above. This method is preferred because with the group function in * you are either ringing every phone in the department driving you crazy when there are many phones, or you have some phones ring and call pickup is more complicated. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supply ringing noise to IAX callers
Hi, Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card connected to a TDA200, when a sip user from box 1 calls someone on the tda200 there is no ringing noise just dead silence until the person on the TDA picks up there extensions. Is there a way in thse situations to supply a ringing sound to the call so the user on box 1 doesn't think there is a problem if the phone is ringing at the other end for 20-30 seconds? James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicetronix dtmf
Good day all I got the latest cvs asterisk But when making a call out threw the voicetronix openline4 card the dtmf doens not work I got this in vpb.conf ecsuppthres = 4096 indication = 1 dtmfidd = 3000 ast-dtmf-det=1 relaxdtmf=1 break-for-dtmf=yes Please help Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 404 User Not Found when calling between two X-Lites
I saw in the dump captured by Ethereal that X-Lite received 200(OK) from asterisk after sending INVITE. So I guessed X-Lite registered well. But I got null reply when I ran sip show peer in asterisk console. What is your opinion about that?On Apr 8, 2005 8:43 PM, Rich Adamson [EMAIL PROTECTED] wrote: The configuration for X-Lite in sip.conf: [177209] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw [177210] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid=Jane Smith 5678 host=dynamic ;nat=yes ; X-Lite is behind a NAT router ;canreinvite=no; Typically set to NO if behind NAT disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw ;allow=alaw The 2 X-Litesregistered well with username 177209 and 177210 respectively. When I made a call between them, I got 404 User Not Found message from asterisk.Any idea? X-Lites both run on Microsoft Windows XP Professional. asterisk 1.07 runs on Red Hat Linux 7.3.Need to look at sip show peers to see if they are actually registered.My first guess they are not since you likely need username= secret=parameters in both of the above examples.If they are in fact registered, then what context are both of theseextensions registered in, and what does that context look like inextensions.conf?___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I exit from asterisk console without stopping asterisk?
If the answer is yes: a) how can I do that? b) how can I restart an asterisk console? Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Re: International callback strategies
Then, I realised a spent lot of time thinkin about this solution. Other option is that you put a prepaid calling card platform in Russia. I saw in CEBIT some russian companies selling prepaid calling cards. In order to give access to your customers without them to know where is the platform, you can also sell them dialers. Dialers call platform and hence people won't know where it is located. Maybe this goes beyond scope but it is workth knowing such solutions exist Selon Adam Goryachev [EMAIL PROTECTED]: On Sun, 2005-04-10 at 15:39 -0700, snacktime wrote: On Apr 10, 2005 3:17 PM, Hakem Taourchi [EMAIL PROTECTED] wrote: 2-) You can create DID system. That is, you buy 1000 DID, and each customer has got a dedicated US did. So when your Callback Systems receive call on DID 101 (without hang up), the callback system knows upfront who to callback; This seems to be the solution that will work the best, although for a small overhead for the did's. Can you get russian DID's routed to you via VoIP?? Then your russian users call a local number to get your system, and then can use disa or astcc or something to make the destination call to USA?? PS, unless of course you can't get DID locally in russia, then maybe you can setup small PC with one/two analog lines and decent internet in someone's home you know (mother/brother/something) ?? Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX calls between asterisk boxes works 1 way only
James Bean wrote:
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'
Error, so I obviously missed something and can someone smack me upside
the head and point out my error.
snip
Just had this happen a couple of minutes ago on our test boxes. You need to
double-check that the Box2's username/password, as specified on Box 1, is
entered properly in Box2's diaplan when dialing to Box 1.
e.g.
Box1 iax.conf
=
[box2]
username=box2
secret=box2secret
Box2 dialplan
=
exten = 777,1,Dial(IAX2/box2:[EMAIL PROTECTED]/${EXTEN})
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[Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS
Hullo :) I've been trying to use a stable 1.0.7 codebase against the patches at http://www.lusyn.com/asterisk/patches.html - but am having no joy. Even if I copy-paste the instructions on that site verbatim, everything compiles perfectly, but simply no incoming number is received. If I then go back to a CVS checkout (even including make clean, make install...) that I did at the end of February, everything works as it did before. (hurrah!) Does anyone know what might have changed before I start wading through the CVS logs for chan_zap.c ? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP UA behind NAT and REINVITE ???
Title: Message My understanding (by no means definitive): You need a solution to the NAT problem for the audio stream. STUN will help with non symmetric NAT but not with symmetric NAT so it's not a complete solution. If you have UAs behind symmetric NAT you will need Asterisk or an RTP proxy in the middle of the call. Regards Cameron - Original Message - From: William M. Sandiford To: asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 7:57 AM Subject: [Asterisk-Users] SIP UA behind NAT and REINVITE ??? Hello: I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? Original Call Made SIP UA 1-- NAT FIREWALL ---Asterisk -- SIP UA 2 Then REINVITE occurs and SIP UA 1-- NAT FIREWALL SIP UA 2 Is this possible? Will using a STUN server help this at all? I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want. I want * out of the audio path) I have even tried putting the private IP of SIP UA 1 in the DMZ of the NAT Firewall and still no luck. Any Suggestions??? Bill --No virus found in this outgoing message.Checked by AVG Anti-Virus.Version: 7.0.308 / Virus Database: 266.9.4 - Release Date: 4/6/2005 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aculab
Hello, on http://www.voip-info.org/wiki-Aculab it has been said, that there is a Aculab card, which works with Asterisk. Two questions: 1. Which card is this? 2. How do I configure it with Asterisk / Linux? If anybody has any experiences regarding this, I would very much appreciate to get some more information on howto use it with Asterisk. Regards Jochen -- Jochen Witte [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR and TDS
Hi, I wantto use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds but when I compile, I've got this message Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"1.0.6-BRIstuffed-0.2.0-RC7k\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -fPIC -c -o cdr_tds.o cdr_tds.ccdr_tds.c: In function `mssql_connect':cdr_tds.c:415: `TDSCONNECTINFO' undeclared (first use in this function)cdr_tds.c:415: (Each undeclared identifier is reported only oncecdr_tds.c:415: for each function it appears in.)cdr_tds.c:415: `connection' undeclared (first use in this function)cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect'/usr/include/ctype.h: At top level:cdr_tds.c:71: warning: `connect_time' defined but not usedmake[1]: *** [cdr_tds.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr'make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call forwarding and parking
Hi !
What is wrong with my dial plan?
I can't get my call forwarding and parking to work.
Do I need to edit more config files?
Thore
extensions.conf :
[general]
static=yes
writeprotect=no
[macro-dialout]
; ${ARG1} CIDNAME
; ${ARG2} Device
; ${ARG3} Num
; ${ARG4} SIP EXT
exten = s,1,SetCIDName(${ARG1})
exten = s,2,Dial(${ARG2}${ARG3}${ARG4},,t)
exten = s,3,Playback(invalid)
exten = s,4,Hangup
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto
102
exten=s,2,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten=s,3,Dial(${ARG2},20) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto
105
exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
[globals]
[apps]
; Unconditional Call Forward
exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.,2,Hangup
exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten = #21#,2,Hangup
; Call Forward on Busy or Unavailable
exten = _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten = _*61*X.,2,Hangup
exten = #61#,1,DBdel(CFBS/${CALLERIDNUM})
exten = #61#,2,Hangup
[iconnect]
exten = _47XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)
exten = _1XXX.,1,Dial(Sip/iconnect/${EXTEN},120,t)
[outgoing-40]
include = apps
include = parkedcalls
exten = _820X,1,Hangup
exten = _,1,Dial(Sip/33297540/${EXTEN},120,t)
exten = _,2,Congestion
exten = _820X,2,Congestion
[outgoing-45]
include = apps
include = parkedcalls
exten = _820X,1,Hangup
exten = _,1,Dial(Sip/voip/${EXTEN},120,t)
exten = _,2,Congestion
exten = _820X,2,Congestion
[local]
include = apps
include = parkedcalls
exten = 101,1,Dial(Sip/101,120)
exten = 102,1,Dial(Sip/102,120)
exten = 201,1,Dial(Sip/201,120)
[dialout-40]
include = outgoing-40
include = local
include = apps
include = parkedcalls
include = iconnect
[dialout-45]
include = outgoing-45
include = local
include = apps
include = parkedcalls
features.conf:
[general]
parkext = 700
parkpos = 701-720
context = parkedcalls
parkingtime = 60
;transferdigittimeout = 3
;courtesytone = beep
adsipark = yes
pickupexten = *8
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Re: [Asterisk-Users] secret/username - what does it really do?
Username and secret in sip.conf are the credentials for the sip user. Any sip UA can then connect to Asterisk using those details and will ring when extension 176 is dialled. Look at sip.conf on the wiki. Regards Cameron - Original Message - From: Don Murray [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 1:02 PM Subject: [Asterisk-Users] secret/username - what does it really do? Hello all, I am working with an AAH installation and Polycom IP 500 phones. Phones are now working and I'm just trying to fine tune what settings I need in my extension and sip .conf files. I have AMP installed obviously (its AAH) but I am finding that I will probably not use the AAH extension set up panel as (a) it isn't phone maker aware (I need to set up stuff particular to Polycom) and (b) it doesn't import manual changes (I can edit sip_additional.conf and AMP will not reflect the manual changes) and (c) it doesn't have all the fields I need (I'm using fixed IP rather than DHCP) Anyway, no big deal, just have to do it myself :) Anyway, if I set up, say, extension 176, via AMP I get the sip and extension entries shown below. What I would like to know is what is the username and secret fields really used for? So far my guess is they identify the mailbox for the voicemail.But I am really not sure and all the documentation I can find on the Wiki about this just gives examples of their use but doesn' treally lay out what they are there for. I've checked the asterisk handbook but it didn't help me as I didn't understand the consequence of the definitions given... I guess because I'm a SIP newbie. We have a asterisk setup that is deep behind a firewall and I am not security conscious at all... the only thing on this network will the asterisk box and the phones. Can I just do away with these values? Thanks for any hints. Don sip.conf [176] username=176 type=friend secret=1766 qualify=no port=5060 pickupgroup= nat=never mailbox= host=dynamic dtmfmode=inband disallow= context=from-internal canreinvite=no callgroup= callerid=Jar Jar Binks 176 allow= [ext-local] exten = 176,1,Macro(exten-vm,176,176) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aculab
Jochen, Recently I contact Aculab in UK about that and They asked me to call Digium Sales. I called Digium Sales and they told me that nothing is confirmed yet about a deal between Aculab and Digium. Maybe something changed Isamar On Mon, 11 Apr 2005, Jochen Witte wrote: Hello, on http://www.voip-info.org/wiki-Aculab it has been said, that there is a Aculab card, which works with Asterisk. Two questions: 1. Which card is this? 2. How do I configure it with Asterisk / Linux? If anybody has any experiences regarding this, I would very much appreciate to get some more information on howto use it with Asterisk. Regards Jochen -- Jochen Witte [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to turn off automatic pick up for Incomingcalls A@H v0.6
Look up the answer command on the wiki. Regards Cameron - Original Message - From: Min Hwan Chang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 10:20 AM Subject: [Asterisk-Users] How to turn off automatic pick up for Incomingcalls [EMAIL PROTECTED] v0.6 I currently use another PBX system which takes care of VM. Is there a way to prevent [EMAIL PROTECTED] v0.6 from picking up Incoming calls? I'd still like to dial out from Asterisk (I have IAX trunking on). Is there a way to do this? My knowledge of the Extensions.conf is limited. I'm using [EMAIL PROTECTED] v0.6. so the conf files were automatically generated and I'm not sure what I should be deleting... or adding. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stand alone Voice Mail
Install Asterisk at home which includes AMP. This will allow you to configure SIP and voicemail using a web browser. Couldn't be easier. Regards Cameron - Original Message - From: Michael D Schelin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 08, 2005 1:45 PM Subject: [Asterisk-Users] stand alone Voice Mail Hello everyone, I need to configure a stand alone Voice mail box. Calls will come in via sip. I have read and read until my eyes hurt for 2 weeks now. Can someone email me the basic config files needed to do this. The examples are overly complicated. I just need a simple basic configurations without all the clutter. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS
On Monday 11 April 2005 10:06, Gavin Hamill wrote: Hullo :) Bah, I got bitten by my own hacks. Things are now working much better than before :) I'd forgotten that I'd commented out the line: if (p-use_callerid p-cid_start == CID_START_USEHIST) in my previous CVS version, and this made CID work. The source of the confusion was I'd put the three 'usecallerid' commands in zapata.conf AFTER a 'channel=1' statement, so they were getting lost... I've moved them up the config file and now not only does incoming CID work, but the phone is answered immediately rather than waiting for the Bellcore CID to not find any data :) Another happy ending :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I exit from asterisk console without stopping asterisk?
To exit from the console type quit To restart the console: type asterisk -vcr (that's several v's in front of the cr. The more v's you put, the greater ther verbosity. I think the max is 10) On Apr 11, 2005 3:30 AM, Abraham WEI [EMAIL PROTECTED] wrote: If the answer is yes: a) how can I do that? b) how can I restart an asterisk console? Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Direct Broadband connection of ip phone to LiveVoip?
Does anyone know if it is possible to connect say Grandstream ip phone directly to LiveVoip? How to setup the phone? Any help will be appreciated! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 07/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Username containing an @
Hi, I have a problem to register with my provider, because my username is myphonenumber@provider's domain. Thus my registry line contains a double @ sign and everything is parsed incorrectly. How can I quote the username to ignore the first @ ? cu chrisb. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PTSN POTS Differences SOLVED
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten = _1NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _1NXXNXX,2,Macro(outisbusy); No available circuits
exten = _9.,1,Macro(dialout-trunk,1,w${EXTEN:1})
exten = _9.,2,Macro(outisbusy); No available circuits
exten = _NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXXNXX,2,Macro(outisbusy); No available circuits
exten = _NXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXX,2,Macro(outisbusy); No available circuits
This fixed the issue. Thanks again. How did you know that?
20+ years of extensive experience in telephony engineering, experience
with Qwest switches that are slow (not all), and a good guess. ;)
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RE: [Asterisk-Users] Supply ringing noise to IAX callers
Whooppss after research for several hours before posting, another asterisk user passed on the answer to me. Add ,r to the Dial string over the E1 to hear the ringing on the line. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Bean Sent: Monday, 11 April 2005 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Supply ringing noise to IAX callers Hi, Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card connected to a TDA200, when a sip user from box 1 calls someone on the tda200 there is no ringing noise just dead silence until the person on the TDA picks up there extensions. Is there a way in thse situations to supply a ringing sound to the call so the user on box 1 doesn't think there is a problem if the phone is ringing at the other end for 20-30 seconds? James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX calls between asterisk boxes works 1 way only
Sorry again sorted it out, the [definition] has to be the same as the
username or it doesn't work, well for me anyway.
:-)
Gotta reasearch a few extra hours and play a bit more before I post I
think.
Sorry guys and girls.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Monday, 11 April 2005 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX calls between asterisk boxes works 1 way
only
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07 NOTICE[2157]: chan_iax2.c:6573 socket_read: Rejected
connect attempt from 192.168.254.100, who was trying to reach '690@'
Error, so I obviously missed something and can someone smack me upside
the head and point out my error.
Please assume that the passwords are correct in the files :-).
Configurations are attached of each box:
Box 1
iax.conf
[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=default
callerid=Guest IAX User
[salisbury]
type=friend
host=192.168.254.100
username=northbuild
secret=password
context=voip
permit=192.168.254.100
extensions.conf
[global]
PSTNLine=Zap/g1
AnalogPhone=Zap/g2
[pstn]
exten = s,1,SetMusicOnHold(random)
exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t)
exten = s,4,VoiceMail(u690)
exten = s,5,Hangup
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 098,1,WaitMusicOnHold(45)
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
exten = 1690,1,VoicemailMain,s690
exten = 1691,1,VoicemailMain,s691
[outgoing]
exten = _9X.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9X.,2,Congestion()
exten = _9X.,3,Hangup
[sip]
exten = 690,hint,SIP/snom-james
exten = 691,hint,SIP/bt-karen
exten = 690,1,SetMusicOnHold(random)
exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten =
690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691
exten = 691,103,Voicemail,b691
include = internal
include = outgoing
include = parkedcalls
include = voip
[voip]
exten = _1XX,1,Dial(IAX2/james:password@192.168.254.100/${EXTEN})
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
-
Box 2
iax.conf
[general]
bindport=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=default
callerid=Guest IAX User
[dixst]
type=friend
host=192.168.50.1
username=dixst
secret=password
context=e100p
permit=192.168.50.1
[james]
type=friend
host=192.168.69.1
username=james
secret=password
context=e100p
permit=192.168.69.1
extensions.conf
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[e100p]
exten = _1XX,1,Dial(Zap/g1/${EXTEN})
exten = _93X.,1,Dial(Zap/g1/${EXTEN})
exten = _9073X.,1,Dial(Zap/g1/${EXTEN})
include = dialstring
include = voip
[voip]
exten = _65X,1,Dial(IAX2/dixst:password@192.168.50.1/${EXTEN})
; DixSt Redcliffe Ext
exten = _66X,1,Dial(IAX2/scarb:password@192.168.60.1/${EXTEN})
; Scarborough
exten = _69X,1,Dial(IAX2/northbuild:password@192.168.69.1/${EXTEN})
; James Home
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Re: [Asterisk-Users] How to turn off automatic pick up for Incoming calls A@H v0.6
I currently use another PBX system which takes care of VM. Is there a way to prevent [EMAIL PROTECTED] v0.6 from picking up Incoming calls? I'd still like to dial out from Asterisk (I have IAX trunking on). Is there a way to do this? My knowledge of the Extensions.conf is limited. Go in AMP, click on Maintenance then Config Edit Click on zapata-channels.conf and locate the channel that is your incoming line. For the context of this line, put context=from-pstn-noanswer. Then, go in extensions_custom.conf and define that context like this : [from-pstn-noanswer] exten = s,1,Wait,2 ; Wait 2 seconds, to get callerid exten = s,2,Hangup This will let your CDR grab the caller ID, then it will hangup, so it won't answer the line. If you don't want the Caller ID, just define it like this : [from-pstn-noanswer] exten = s,1,Hangup hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P power supply
I've a problem with a TDM400P digium card. My box has no molex connectors for power supply. Simply has no any power connector, because is not a normal PC) And I need to know if i can use a external supply. But I've several questions: 1.- Are both circuits (PCI-power and Phone-line-power) electrically separated? 2.- A little voltage difference can create an undesired internal current? 3.- What are the current needs for this supply? I need the power supply because I want to use both FXS and FXO ports. And I can't use a Y cable, because I've no molex connectors. Been discussed several times before and you should have found the answer using google. The TDM connector is only used for the fxs modules, and then only the +12 volt lead on that connector (and ground) is actually wired to anything on the TDM board. So, there is no conflict with internal system voltages. Yes you can use an external 12 volt power supply. The 12 volts is only used on the card to generate ringing voltage to the fxs modules. No ringing, no significant current draw. Just about any 12 volt supply should do, however I think I'd be looking for one that is at least somewhat regulated. No other idea on the power supply specs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CDR and TDS
In article [EMAIL PROTECTED], David Masure [EMAIL PROTECTED] wrote: I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds but when I compile, I've got this message Does anyone have the same problem and/or know how to solve it ? Looks like you probably have version 0.63 of FreeTDS. That version is incompatible with cdr_tds, but if you're using ODBC you should be fine. However, you will need to tell the Makefile not to compile cdr_tds, by commenting out the following two lines in cdr/Makefile: MODS+=$(shell if [ -f /usr/include/tds.h ]; then echo cdr_tds.so; fi) MODS+=$(shell if [ -f /usr/local/include/tds.h ]; then echo cdr_tds.so; fi) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 404 User Not Found when calling between two X-Lites
I saw in the dump captured by Ethereal that X-Lite received 200(OK) from asterisk after sending INVITE. So I guessed X-Lite registered well. But I got null reply when I ran sip show peer in asterisk console. What is your opinion about that? If sip show peers does not show your xlite boxes, then they aren't registered. You really need to figure out why they aren't registering before attempting any calls, etc. You might consider using sip debug and sip no debug to see what is actually going on. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Fax to Email
This has already been answered...but I can't find it... Has anyone set up multiple fax lines in asterisk... Fax Extension #1 goes to email1 Fax Extension #2 goes to email2 ETC... In other words, I want to be able to give numerous users each a virtual fax machine.. Bill ; Assumes entry is DID # or extension number [context-incoming] exten = some_did,1,NVFaxEmail([EMAIL PROTECTED],Someone) exten = some_other_did,1,NVFaxEmail([EMAIL PROTECTED],Someone2) You could use NVFaxDetect first to check for the presence of the fax. This sample requires SpanDSP and NVFaxEmail. Alternatively, you could use SpanDSP, RxFax, and a different AGI script or app. Not sure what pstn interface the OP was using, but if its a digium TDM analog card, SpanDSP will not function correctly due to frame slips in most systems. SpanDSP is working great on my analog TDM card. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: no ring on inbound SIP calls
On Sun, 10 Apr 2005, Eric Wieling wrote: No. r instructs Asterisk to provide a fake ringback tone. If you need r then something is seriously wrong. Asterisk will always provide rinback tones when it thinks it should. For PRI channels you may need it if the equipment at the oher end does not provide in band audio. Also, Asterisk does not synthezise in band audio progress information when bridging pri-sip at least. This is despite Asterisk sending a progress message with in band progress available. It seems quite normal to need r on internal extension (going to a pbx over pri or to some sip phones). Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM02B on 2 a/b ports of a PBX not working... help
I own elmeg C46e PBX (ISDN PBX with 6 a/b ports). I connected two of these ports to a TDM02B installed on a Slackware 9.1/ Asterisk 1.0.2 and I bought a Cisco ATA186 to connect two analog telephone sets two floors down. Each a/b port is assigned only to one ATA port. The problem is that after the first call, the Digium card fails to disconnect the lines and I keep getting the busy signal. What is more strange is that when I connect the same equipment to two POTS lines of the telco, all run smoothy. Any ideas anyone? How should I setup my line (eg as fxs_ks or something else?) Dimitris Kouimintzis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Attended/Supervised transfer features.conf
Hi all, We were willing to try the SIP Attended/Supervised transfer with * realease 1.0-7. From the wiki´s feature.conf config page we found that a special section called featuremap had to be added to the config: [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer We made that changes but upon pressing *2 nothing happens, neither with #1 for the blind transfer. The blind transfer is working as it defaults in *, with # plus extension. We tried to unload and reload res_features module but with no luck as it says that the user count is 1. After some examination at chan_sip.c, we found the supervised transfer code section, but we found nothing on the parsing of the featuremap section. We did find the parsing of the first section of the config file concerning call parking, which does work. Any idea on how to make it work? Thanks to all, Gonchi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Servers and 1 Central Voicemail
MWI works just fine. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPS version 0.79 released
Version 0.79 - 11. April 2005. * Norvegian language added - thanks to Kåre Sundland * German language updated - thanks to Marco Walz * Russian language updated - thanks to dnz63 * Caller ID name added to call buttons when on call FREE Download: http://ipswitchboard.thorben.dk Would you like to help translate IPS into your language? Please click the link below for details. I will add your language as soon as I receive it. http://ipswitchboard.thorben.dk/index.php?option=com_simpleboardItemid=42f unc=showcatcatid=5 IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: * Unattended/attended transfers. * Park calls and retrieve/forward them again. * Organize all your SIP and IAX extensions (automatically retrieved from Asterisk). * Monitor all extensions. * Monitor all queues. * Monitor Agents. * Monitor Parked Calls. * Dynamically log extensions in and out of queues. * Integration with CRM software on the web. * Drop any active call. * Import/Export extensions to/from Asterisk Server DB. * Set Do Not Disturb on Extensions and give a reason. * Speed Dialling. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A101 + Rhino channelbank
Hello, I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with the Rhino fxs chanelbank. Things done: - T1 cross cable = I have carrier, signalling and framnig leds on the channelbank green. - channelbank configuration: t1 - Proto: LOOP Frame: esf Clock: slave Coding: b8zs channels(analog) : Function:A-fxsMode:loop - zaptel.conf span=2,1,0,esf,b8zs fxols=32-55 (i have a span 1 with a digium e1) - zapata.conf signalling=fxo_ls - wanpipe1.conf [devices] wanpipe1 = WAN_AFT, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 2 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= MASTER ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO LBO = 0DB INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 2 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO I already called Sangoma and Rhino support, but after hours of long distance call conversation the problem is still not solved. Finnaly, a guy from Rhino told me that their asterisk expert (which was not avaliable) knows about this problem and that it is that the sangoma driver is not communicating with asterisk. The wanrouter starts ok, after ztcfg I see the channels configured. The problem: i don't have dialtone on phones. Question: When i enter zttoll, if i go to the sangoma span and I make loop then it freezes. Is it normal? If someone has experienced this combination and made it work please give me a sign. Thank you. PS: Felician ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I exit from asterisk console without stoppingasterisk?
exit and asterisk -r - Original Message - From: Abraham WEI To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 11, 2005 3:30 AM Subject: [Asterisk-Users] Can I exit from asterisk console without stoppingasterisk? If the answer is yes: a) how can I do that? b) how can I restart an asterisk console?Best regards,Abe ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
Giles thank you for getting back so quickly, dmesg doesnt output anything, but even if it did, I am not sure that I could recompile the kernel. The server I am using is in a virtual dedicated hosting environment, I do not have access to recompile the kernel, nor can I replace it. The server prevents me from doing so. I do not have access to the real /boot and dont have access as far as I can tell to the .config for the kernel source. (make oldconfig seems to work) After a few more days of tech support, google searches and etc, I have found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course, cat /proc/version doesnt think so!! It thinks I am running Kernel 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source the corresponding rpm which installs without incident. The interesting part is rpm qa kernel doesnt see it L. I even tried to rpm rebuilddb Zaptel appears to compile fine, but when I run modprobe zaptel I get the following: -- /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o was compiled for kernel version 2.4.21-4.0.1.EL while this kernel is version 2.4.20-021stab022.11.777-enterp. /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod zaptel failed - Is there a way to override zaptels kernel check or have linux fool it into thinking the kernel is 2.4.21-4.0.1.EL? thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Wednesday, April 06, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Anyone have any ideas on where I can find the right kernel source? I have look at rpmfind.net and google'd with no avail! You could always download the Vanilla kernel source from http://www.kernel.org and compile a kernel from source. I tend to always use the Vanilla source, it's what everything has been tested against and it tastes better. You should probably print out the dmesg output to help you configure the kernel options prior to compilation so that your hardware is correctly detected. I would also urge you to use a bootloader such as grub or lilo to ensure that you can revert to the original kernel should it panic on boot, I suspect Redhat already uses one of those anyway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Shared call appearances
Has anyone worked out how to get the Shared call appearances working on a SPA841 with Asterisk. Googling found a few people asking the same question last year, but alas no answers. Just wondering if anybody has made the breakthrough in the meantime. craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
Title: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host. Hi, Do you happen to know what VPS system your host uses (e.g. UML, Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as some platforms will allow changes that others will not. -- Henry Owens. On 11/4/05 2:20 pm, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Giles thank you for getting back so quickly, dmesg doesnt output anything, but even if it did, I am not sure that I could recompile the kernel. The server I am using is in a virtual dedicated hosting environment, I do not have access to recompile the kernel, nor can I replace it. The server prevents me from doing so. I do not have access to the real /boot and dont have access as far as I can tell to the .config for the kernel source. (make oldconfig seems to work) After a few more days of tech support, google searches and etc, I have found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course, cat /proc/version doesnt think so!! It thinks I am running Kernel 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source the corresponding rpm which installs without incident. The interesting part is rpm qa kernel doesnt see it L. I even tried to rpm rebuilddb Zaptel appears to compile fine, but when I run modprobe zaptel I get the following: -- /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o was compiled for kernel version 2.4.21-4.0.1.EL while this kernel is version 2.4.20-021stab022.11.777-enterp. /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod zaptel failed - Is there a way to override zaptels kernel check or have linux fool it into thinking the kernel is 2.4.21-4.0.1.EL? thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Wednesday, April 06, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Anyone have any ideas on where I can find the right kernel source? I have look at rpmfind.net and google'd with no avail! You could always download the Vanilla kernel source from http://www.kernel.org and compile a kernel from source. I tend to always use the Vanilla source, it's what everything has been tested against and it tastes better. You should probably print out the dmesg output to help you configure the kernel options prior to compilation so that your hardware is correctly detected. I would also urge you to use a bootloader such as grub or lilo to ensure that you can revert to the original kernel should it panic on boot, I suspect Redhat already uses one of those anyway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
On Apr 6, 2005 10:34 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Anyone have any ideas on where I can find the right kernel source? I have look at rpmfind.net and google'd with no avail!Hi, You're never going to find the kernel source. The reason for this is that your VPS is running under Virtuozzo, which is a commercial software package designed to create virtual servers under one physical server, sharing a common kernel. This means the kernel cannot be upgraded or in any way modified. Regards, Gonzalo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] append # to dial string
John Breeden wrote:
Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how you
can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? -
hex/octal whatever? I'm surprised that I can't find anything
searching the wiki or google.
Try something like this:
exten = _9XXX,1,Dial(Zap/4/${EXTEN}#)
Then you are doing something wrong. The above syntax is correct.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards
[EMAIL PROTECTED] wrote: Hi, How can i implement VAD/DTX using zaptel with asterisk towards PSTN. TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not even a valid idea. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR and TDS
David Masure wrote: Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds but when I compile, I've got this message Does anyone have the same problem and/or know how to solve it ? Update of /usr/cvsroot/asterisk/doc In directory mongoose.digium.com:/tmp/cvs-serv24936/doc Added Files: README.tds Log Message: Add documentation for TDS noting compilation problem on 0.63+ --- NEW FILE: README.tds --- PLEASE NOTE The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. The cdr_tds module is known to work with FreeTDS version 0.62.1; it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug fix releases. The cdr_tds module uses the raw libtds API of FreeTDS. It appears that from 0.63 onwards, this is not considered a published API of FreeTDS and is subject to change without notice. Between 0.62.x and 0.63 of FreeTDS, many incompatible changes have been made to the libtds API. For newer versions of FreeTDS, it is recommended that you use the ODBC driver. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Compile on a virtual dedicated host.
Title: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host. It appears to be Virtuozzo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Sent: Monday, April 11, 2005 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated host. Hi, Do you happen to know what VPS system your host uses (e.g. UML, Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as some platforms will allow changes that others will not. -- Henry Owens. On 11/4/05 2:20 pm, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Giles thank you for getting back so quickly, dmesg doesnt output anything, but even if it did, I am not sure that I could recompile the kernel. The server I am using is in a virtual dedicated hosting environment, I do not have access to recompile the kernel, nor can I replace it. The server prevents me from doing so. I do not have access to the real /boot and dont have access as far as I can tell to the .config for the kernel source. (make oldconfig seems to work) After a few more days of tech support, google searches and etc, I have found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course, cat /proc/version doesnt think so!! It thinks I am running Kernel 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source the corresponding rpm which installs without incident. The interesting part is rpm qa kernel doesnt see it L. I even tried to rpm rebuilddb Zaptel appears to compile fine, but when I run modprobe zaptel I get the following: -- /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o was compiled for kernel version 2.4.21-4.0.1.EL while this kernel is version 2.4.20-021stab022.11.777-enterp. /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod zaptel failed - Is there a way to override zaptels kernel check or have linux fool it into thinking the kernel is 2.4.21-4.0.1.EL? thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Wednesday, April 06, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Anyone have any ideas on where I can find the right kernel source? I have look at rpmfind.net and google'd with no avail! You could always download the Vanilla kernel source from http://www.kernel.org and compile a kernel from source. I tend to always use the Vanilla source, it's what everything has been tested against and it tastes better. You should probably print out the dmesg output to help you configure the kernel options prior to compilation so that your hardware is correctly detected. I would also urge you to use a bootloader such as grub or lilo to ensure that you can revert to the original kernel should it panic on boot, I suspect Redhat already uses one of those anyway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm
On Tue, Apr 05, 2005 at 09:06:33PM +0200, Peter Svensson wrote: A yellow alarm means the remote end is sensing some error condition. Try looking for an error message at the remote end. It may be as easy as a broken cable (where the Hipath does not hear the Asterisk box). The problem is, that the TMS2-Card in the HiPath is not activated, it says, that the line is dead. According to the Siemens-People the Card should activate itself as soon as a signal reaches the card. But it appears, that Asterisk sends no signal. This is what the layout looks like: Asterisk|TE110P - TMS2|HiPath|TMS2 - PSTN The cable is functional and the wiring is correct. But I'm not sure how I must configure the TMS2 card. Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: PTSN POTS Differences SOLVED
Tony, I don't see ${EXTEN} anywhere in the [macro-dialout-trunk] context.
Am I missing something?
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: [EMAIL PROTECTED] (Tony Mountifield)
Organization: Software Insight Ltd., Winchester, UK
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 11 Apr 2005 07:48:19 + (UTC)
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: PTSN POTS Differences SOLVED
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten = _1NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _1NXXNXX,2,Macro(outisbusy); No available circuits
exten = _9.,1,Macro(dialout-trunk,1,w${EXTEN:1})
exten = _9.,2,Macro(outisbusy); No available circuits
exten = _NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXXNXX,2,Macro(outisbusy); No available circuits
exten = _NXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXX,2,Macro(outisbusy); No available circuits
Couldn't you have just put the w in once, in the Dial command that
is inside [macro-dialout-trunk] ?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. Does a great job. My only fear is it doesn't specifically target IAX2 traffic as high priority, but I can modify it later to do so if needed. On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no noticable problems. Along with someone streaming a shoutcast station ( sigh ). The station broke up, but the calls didn't. cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? Thanks for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interface bonding + asterisk
Hi all I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to configure both NICs with bonding enable (bonding miimon=100 mode=1). I know certain features (like load balancing) under a bonded configuration is not understood by some switches, so I configured it using mode=1 (Failover only). The problem I'm having is that, sometimes, calls start fine but then one of the parties loses audio (it could be the caller of the callee who loses audio, there is no pattern). I was wondering if someone has hit the same wall as me. There are people using this server right now, so I haven't tried the no-bonding option as it means downtime. Any help would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How doesone shape RTP?
I agree that Wondershaper is a great script; prior to using it in an office where I set up asterisk, there were some major problems with call quality, but it seems to have helped hugely (the same DSL line is used for both VoIP and everyday 'net usage for seven people - not ideal, but I didn't set the budget :-) ). If you happen to modify it to to prioritize IAX2, drop me a copy! -- Henry Owens. On 11/4/05 3:08 pm, Sean Kennedy [EMAIL PROTECTED] wrote: Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. Does a great job. My only fear is it doesn't specifically target IAX2 traffic as high priority, but I can modify it later to do so if needed. On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no noticable problems. Along with someone streaming a shoutcast station ( sigh ). The station broke up, but the calls didn't. cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? Thanks for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
I used the one posted to this list and for a test did a speedtest.dslreports.com bandwidth test duringa call, no loss in quality. I set ports 1-11024 to RTP in rtp.conf, I dont need 10k ports for that as I have few calls being processed. I also added sip to the queue although that prolly doesnt matter becuase its such a low bandwidth protocol comparitevly speaking. # udp/5060 is SIP tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip dport 506 0 0x match ip protocol 17 0xff flowid 1:0 tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip sport 506 0 0x match ip protocol 17 0xff flowid 1:0 # udp/1-11024 is RTP tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip dport 100 00 0xf670 match ip protocol 17 0xff flowid 1:0 tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip sport 100 00 0xf670 match ip protocol 17 0xff flowid 1:0 On Mon, 2005-04-11 at 07:08 -0700, Sean Kennedy wrote: Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. Does a great job. My only fear is it doesn't specifically target IAX2 traffic as high priority, but I can modify it later to do so if needed. On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no noticable problems. Along with someone streaming a shoutcast station ( sigh ). The station broke up, but the calls didn't. cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? Thanks for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank
Keep on bugging the Sangoma guys, I know they are working on several RBS T1 issues right now(They called me Friday to go over a few things) They just need help from users like you and I to find the bugs in their drivers. Have you tried any other signalling types other than LOOP? MATT--- -Original Message- From: Felician CHELU [mailto:[EMAIL PROTECTED] Sent: Monday, April 11, 2005 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank Hello, I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with the Rhino fxs chanelbank. Things done: - T1 cross cable = I have carrier, signalling and framnig leds on the channelbank green. - channelbank configuration: t1 - Proto: LOOP Frame: esf Clock: slave Coding: b8zs channels(analog) : Function:A-fxsMode:loop - zaptel.conf span=2,1,0,esf,b8zs fxols=32-55 (i have a span 1 with a digium e1) - zapata.conf signalling=fxo_ls - wanpipe1.conf [devices] wanpipe1 = WAN_AFT, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 2 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= MASTER ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO LBO = 0DB INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 2 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO I already called Sangoma and Rhino support, but after hours of long distance call conversation the problem is still not solved. Finnaly, a guy from Rhino told me that their asterisk expert (which was not avaliable) knows about this problem and that it is that the sangoma driver is not communicating with asterisk. The wanrouter starts ok, after ztcfg I see the channels configured. The problem: i don't have dialtone on phones. Question: When i enter zttoll, if i go to the sangoma span and I make loop then it freezes. Is it normal? If someone has experienced this combination and made it work please give me a sign. Thank you. PS: Felician ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
On April 11, 2005 10:08 am, Sean Kennedy wrote: Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. :-) I started out with wshaper and just didn't like it, which is where rc.tc came from. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with X101P
Previously I have posted the same mail but no one answered me...Sorryfor resending the mail.I have bought a Wildcard X101P for my Asterisk PBX. Now I can placeand get calls through the lines/channel. Everything is okay but theproblem is when I call outside through our PSTN line, after fewminutes the connection breaks down. The same thing happens in case ofincoming calls. I have checked my wiring and don't face that problemusing direct connection. Whenever I call using that card, after fewminutes I get a RED Alarm and if I reconnect the line, the Alarm iscleared.Therefore, I cannot continue my conversation through that line. Cananybody help me regarding this problem? Express yourself instantly with MSN Messenger! MSN Messenger Download today it's FREE! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
On Mon, 2005-04-11 at 10:32 -0400, Andrew Kohlsmith wrote: On April 11, 2005 10:08 am, Sean Kennedy wrote: Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. :-) I started out with wshaper and just didn't like it, which is where rc.tc came from. you may want to pull at least the RTP lines I just posted and add them to your rc.tc since that is what I got and tweaked since I use RTP :) -- Trixter http://www.0xdecafbad.com signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo problem
I've got a X100P in a compaq proliant 3000. My system stops taking calls and making calls. I had been getting the FXO PCI Master abort before updating, I am now running a cvs head checkout from a week or so ago. Now I still have the problem but get more error messages: Found a Wildcard FXO: Wildcard X101P Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) FXO PCI Master abort wcfxo: Out of space to write register 05 with 02 wcfxo: Out of space to write register 05 with 03 wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a Any solution? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-841 Phone Review
I use a headset w/out any problems, except for if my cell phone is close by and rings. Otherwise, volume is ok and no humming. Could it be your headset? At 01:56 PM 4/10/2005, you wrote: Just make sure you don't have a cordless or cell phone near by or the headset jack will receive a considerable amount of interference into your conversation (when NOT using a headset). Also don't even try using a headset... volume is low and there is a loud humming noise. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From OH323 to SIP or OH323 without gatekeeper
Joe S [EMAIL PROTECTED] writes: Hi, I am new with asterisk. I was wondering if there is a way to call a OH323 user or SIP user using Netmeeting/SJPhone with H323 as the default protocol without having a gatekeeper. I can make a call from SIP to OH323 by specifying it in the extensions.conf file, like: exten=1001, 1, Dial(OH323/10.10.10.1) so I was wondering if there was a way to call from OH323 to SIP or OH323. Sure. Just specify in oh323.conf the context where incoming calls should go. That context then can include dial statements for any protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to setup dial plans. Finally, instruct your H323 phone to use asterisk as a gateway resp. proxy, not a gatekeeper. Any calls will then go through asterisk, and to the context you specified. I'm doing that with Gnomemeeting all the time, and it works without problems. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxo problem
I'm having similar issues using an X100P Ambient Chipset Clone Card any ideas? Regards, Sahil Gupta VoiceValley On Mon, 11 Apr 2005, Dave Weis wrote: I've got a X100P in a compaq proliant 3000. My system stops taking calls and making calls. I had been getting the FXO PCI Master abort before updating, I am now running a cvs head checkout from a week or so ago. Now I still have the problem but get more error messages: Found a Wildcard FXO: Wildcard X101P Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) FXO PCI Master abort wcfxo: Out of space to write register 05 with 02 wcfxo: Out of space to write register 05 with 03 wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a wcfxo: Out of space to write register 05 with 0a Any solution? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Revision question.
Sorry for the initial no subject line. Was in a hurry when I typed this and somehow missed putting it in. Please accept my apologies On Mon, 11 Apr 2005 10:54:30 -0400 Robert Webb [EMAIL PROTECTED] wrote: Good morning all.. I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intercom with Aastra 480e?
Hello list, I have been successful in setting up my first * box with a pair of x100ps, Cisco 7960, and a Digium iAXy. I would like to incorporate an Aastra 480e using my iAXy and ADSI. I want to be able to answer phone calls with my 7960 in the back of the house and park the call, then in turn call the intercom on the 480e in the front (using two way audio) to announce that there is a call that needs to be picked up on 701. Also, by using the Aastra 480e, can I see my Zap line status to see what lines are available and also if extensions are in use? Thanks in advance. B. Lacey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manipulate Asterisk Database from manager?
Hi, Is there anyway to manipulate the asterisk internal database from the manager (the one you can telnet to)? And if so.. how does one do it? (ie for enabling call forwarding, etc) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why 's' doesn't take over unknown extension in context ?
Hi,
I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk)
I have context 'from-isdn' :
[from-isdn]
exten = s,1,Wait,2
exten = s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number: ${CALLERIDNUM})
exten = s,3,SetCIDName(From ISDN: ${CALLERIDNUM})
exten = s,4,SetCIDNum(0${CALLERIDNUM})
exten = s,5,AGI,callerid_lookup.agi
exten = s,6,NoOp(After callerid_lookup.agi: ${CALLERID}: Name:
${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten = s,7,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = s,8,DBdel(DYNAMIC/${CALLERIDNUM})
exten = s,9,Background(custom/aa_1)
exten = s,10,Wait,5
exten = s,11,Dial(Local/[EMAIL PROTECTED]/n)
exten = s,108,Goto(from-pstn,s,1) ;
exten = 99,1,Goto(s,1) ;
Now if there is no line 99 on incoming call I get :
-- Extension '99' in context 'isdn-incoming' from '041461620' does
not exist. Rejecting call on channel 0/1, span 1
Why doesn't extension 's' get started if extension 99 is unknown in
context from-isdn?
Thanks in advance,
regards,
Rob.
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[Asterisk-Users] timed Loop
I need to make a time loop in the Extensions.conf. I want it to play a file every 5 minutes on a call. If I can't use wait because it ignores all audio. Anyone have any suggestions? Regards, Chris___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
I was following a discussion on this list about the TDM400P revisions. It is my understanding that the current revision that one should have is the Rev. H and not the E/F. I have not yet been able to verify the rev stamped on the board, but zaptel is reporting that I have the Rev. E/F. I just bought this card in January direct from Digium and was wondering if I got the wrong Rev. somehow?? I have been having some intermittent problems but only thought it was my setup. I'm not sure when they came out with the Rev H one. If you look back at the archives over the last year, you'll see several people that have had problems and several more that have not had any problems at all. There does not seem to be any common ground for those that have had problems. Gut feeling (and some rather general comments) tend to suggest the issue is associated with the pci bus, and possibly something to do with the TigerJet pci controller on the card. Best guess is that it has something to do with pci bus timing issues and that probably is somewhat dependent on the exact motherboard in use. Someone posted a note a few weeks ago that essentially said, if your tdm card goes out to lunch (every week or two), dump the tdm registers, and if their all zero's (or 0xff's forget which), then the card should be replaced. The Rev H card _does_ have some additional components on it close to the TigerJet chip, and the fxo modules are now marked as x100 (which they were not marked on the originals). So, something in the design has changed. Hopefully, its an improvement. :) I won't know for another two weeks or so. Best bet is to call digium support and let them walk you through it. It only took about 30 minutes for me last week, and after I described my problem they offered to RMA it without saying anything more, and without logging into my system. Must have been pretty obvious/familiar. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?
I have a home user for asterisk that is not ready to let asterisk manage the entire dialplan ... he's still got an answering machine on the outside line and has this in the [incoming] context for that line: exten = s,1,Wait(300) exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 He does this so the answering machine can answer the phone when he's busy or not available, but can still dial out on asterisk. The Wait(300) prevents asterisk from answering before the answering machine does. Crazy, huh? Anyway ... he wants to be able to pickup an incoming call during the Wait(300). Can this be done? Thanks, lane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting CVS HEAD
Hi, I want to download the CVS HEAD version. Any one can show how to get this version ? Is the version from: http://www.asterisk.org/index.php?menu=download the CVS Head version? How I can check if my version is CVS HEAD or not? Best Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why 's' doesn't take over unknown extension incontext ?
I think it is i you want, s is the start for a context, meaning anything
coming in through that context will start there, i is invalid, and fires
if an invalid extension is keyed in that context.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Rozman
Sent: Monday, April 11, 2005 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Why 's' doesn't take over unknown extension
incontext ?
Hi,
I always thought that if there is no called extension in context, then 's'
extension is started (I use latest bristuffed Asterisk)
I have context 'from-isdn' :
[from-isdn]
exten = s,1,Wait,2
exten = s,2,NoOp(ISDN call from outside ${CALLERID}: Name: ${CALLERIDNAME},
Number: ${CALLERIDNUM})
exten = s,3,SetCIDName(From ISDN: ${CALLERIDNUM})
exten = s,4,SetCIDNum(0${CALLERIDNUM})
exten = s,5,AGI,callerid_lookup.agi
exten = s,6,NoOp(After callerid_lookup.agi: ${CALLERID}: Name:
${CALLERIDNAME}, Number: ${CALLERIDNUM})
exten = s,7,DBget(temp=DYNAMIC/${CALLERIDNUM})
exten = s,8,DBdel(DYNAMIC/${CALLERIDNUM})
exten = s,9,Background(custom/aa_1)
exten = s,10,Wait,5
exten = s,11,Dial(Local/[EMAIL PROTECTED]/n)
exten = s,108,Goto(from-pstn,s,1) ;
exten = 99,1,Goto(s,1) ;
Now if there is no line 99 on incoming call I get :
-- Extension '99' in context 'isdn-incoming' from '041461620' does
not exist. Rejecting call on channel 0/1, span 1
Why doesn't extension 's' get started if extension 99 is unknown in
context from-isdn?
Thanks in advance,
regards,
Rob.
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Re: [Asterisk-Users] Getting CVS HEAD
If You do a checkout in CVS without specifying a version (as shown on the referenced site) You will allways get the HEAD (means the most recent) branch. Jochen Am Montag, den 11.04.2005, 10:27 -0500 schrieb Guillermo Salas M: Hi, I want to download the CVS HEAD version. Any one can show how to get this version ? Is the version from: http://www.asterisk.org/index.php?menu=download the CVS Head version? How I can check if my version is CVS HEAD or not? Best Regards, -- Jochen Witte [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simulated dialtone like in other PBX
I have a question about dial tone as well, when calling a company typically they have an answering system, i.e. press 4 for Bill etc. I am using a diax soft-phone and have been unable to get the receiving system to forward me on. Is there a feature change in Asterisk that needs to be enabled or is it a problem with diax? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, February 20, 2005 11:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Simulated dialtone like in other PBX Thx Sergey!! Ill give it a try -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Kuznetsov Sent: Domingo, 20 de Febrero de 2005 07:34 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simulated dialtone like in other PBX Easy as piece of cake. Remove ignorepat=9 add: exten = 9,1,DISA(no-password|my_outbound_context) [my_outbound_context] exten = NXX, 1, blah-blah-blah All the Best! Sergey. Peter Svensson wrote: On Sun, 20 Feb 2005, Anton Krall wrote: Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? I'm not sure I understand your question. Do you want to be able to hit 9 and get a an outside line with dialtone? Just add an extension to do that. For isdn you need to enable overlap dialing. Or do you want Asterisk to provide a dialtone after the user have hit 9 as the first digit of a number? User the ignorepat option in the dialplan. Or do you want Asterisk to provide a _different_ dialtone after the user have hit 9 as the first digit of a number? This may be possible, but I think some hack may be needed. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?
I have a home user for asterisk that is not ready to let asterisk manage the entire dialplan ... he's still got an answering machine on the outside line and has this in the [incoming] context for that line: exten = s,1,Wait(300) exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 He does this so the answering machine can answer the phone when he's busy or not available, but can still dial out on asterisk. The Wait(300) prevents asterisk from answering before the answering machine does. Crazy, huh? Anyway ... he wants to be able to pickup an incoming call during the Wait(300). Can this be done? Sure, but not with the stuff shown above. Just use something like: [inbound-home] exten = s,1,Dial(SIP/3000,15) Asterisk won't answer the inbound call unless someone picks up the sip phone. If no one picks it up, it stops ringing after 15 seconds. The bridged answering machine does its thing whenever it wants to. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Attended/Supervised transfer features.conf
On 09:17, Mon 11 Apr 05, Gonchi Mateos wrote: Hi all, We were willing to try the SIP Attended/Supervised transfer with * realease 1.0-7. From the wiki?s feature.conf config page we found that a special section called featuremap had to be added to the config: [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer We made that changes but upon pressing *2 nothing happens, neither with #1 for the blind transfer. The blind transfer is working as it defaults in *, with # plus extension. We tried to unload and reload res_features module but with no luck as it says that the user count is 1. After some examination at chan_sip.c, we found the supervised transfer code section, but we found nothing on the parsing of the featuremap section. We did find the parsing of the first section of the config file concerning call parking, which does work. This only works on CVS version of *, not the stable 1.0 -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with X101P
Some questions: What country are you in? Is there anything else connected to the line from the PSTN? It sounds like you have a marginal condition, such as insufficient loop current perhaps. Do have any features, such as call waiting, on the line? Do you know how far you are from the central office? Do you have another line you can switch to and try the same card? Does the Red alarm occur at the moment the call is disconnected, or afterward? Regards Scott Stingel www.evtmedia.com Yusuf Iqbal wrote: Previously I have posted the same mail but no one answered me...Sorry for resending the mail. I have bought a Wildcard X101P for my Asterisk PBX. Now I can place and get calls through the lines/channel. Everything is okay but the problem is when I call outside through our PSTN line, after few minutes the connection breaks down. The same thing happens in case of incoming calls. I have checked my wiring and don't face that problem using direct connection. Whenever I call using that card, after few minutes I get a RED Alarm and if I reconnect the line, the Alarm is cleared. Therefore, I cannot continue my conversation through that line. Can anybody help me regarding this problem? // Express yourself instantly with MSN Messenger! MSN Messenger http://g.msn.com/8HMAEN/2728??PS=47575 Download today it's FREE! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: PTSN POTS Differences SOLVED
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Tony, I don't see ${EXTEN} anywhere in the [macro-dialout-trunk] context.
Am I missing something?
It's probably ${ARG2}.
When you call Macro(name,1,${EXTEN}), say for extension 1234, then the
macro [macro-name] gets called with ${ARG1} containing 1, and ${ARG2}
containing 1234.
Somewhere in [macro-dialout-trunk] you probably have the command:
Dial(whatever/something/${ARG2},options)
If that's true, you just need to put the w before ${ARG2}.
I've never seen [EMAIL PROTECTED], so if the above doesn't match what you
have, perhaps you could post the contents of [macro-dialout-trunk].
Cheers
Tony
Robert Andrew Keller
Ferndale School District #502
[EMAIL PROTECTED]
360-383-9228 PH.
360-383-9218 FAX
Paving the way for tomorrows genius.
From: [EMAIL PROTECTED] (Tony Mountifield)
Organization: Software Insight Ltd., Winchester, UK
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 11 Apr 2005 07:48:19 + (UTC)
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: PTSN POTS Differences SOLVED
In article [EMAIL PROTECTED],
Robert Keller [EMAIL PROTECTED] wrote:
Thanks Rich, I wasn't sure where to find that context. I found the outbound
context in the extensions_additional.conf and added w's in the following
manner:
[outrt-001-Out1]
include = outrt-001-Out1-custom
exten = _1NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _1NXXNXX,2,Macro(outisbusy); No available circuits
exten = _9.,1,Macro(dialout-trunk,1,w${EXTEN:1})
exten = _9.,2,Macro(outisbusy); No available circuits
exten = _NXXNXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXXNXX,2,Macro(outisbusy); No available circuits
exten = _NXX,1,Macro(dialout-trunk,1,w${EXTEN})
exten = _NXX,2,Macro(outisbusy); No available circuits
Couldn't you have just put the w in once, in the Dial command that
is inside [macro-dialout-trunk] ?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Linux Asterisk
As you are a new Linux and asterisk user you best path is to use a Linux Distribution that is easy to install and setup. I have heard that Mandrake is very good, but for me I like Fedora 2/3 from Red hat. You will need an OS that has clear documentation in the form of books and a well supported user community that you are comfortable with. That is the quick answer, the better answer is: You need to first decide which software you want to run on your server, this will then tell you which hardware you should run one, motherboard, graphics card, and most importantly which LAN Ethernet card and which Telephone interface card. The you need to know which Linux OS will support and has the drivers for all the equipment that you. Step 1. Decide what features you need as to telephone connections to the box. ISDN, BRI, PRI, T1, SS7, FXS, FXO. 2. Then find the software the will support your connection needs. 3. Then find the OS that will allow both of these to connect together. It is very important to have the drivers supported by the OS you are using. Ask or google for that sort of thing. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Dwyer Sent: Sunday, April 10, 2005 11:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Pbx Subject: Re: [Asterisk-Users] Linux Asterisk Hi, I think the correct answer to this question is whatever you are most comfortable with. I also use Debian Sarge and find it to be great. I'm also a FreeBSD user. Perhaps the best thing to do would be to install one distro and have a play and see if you like it. If you don't, try something else until you find something you are comfortable with. Cheers, -Shaun Matteo Brancaleoni wrote: Hi, Il giorno gio, 07-04-2005 alle 05:19 -0400, Asterisk Pbx ha scritto: I am thinking in implementing asterisk into my buisness. I heard all sorts of good things about it. The question im asking my self is what linux distribution is best to use? Do you know what distribution they use for their asterisk training? Please search the ML. this question has been asked as many times as the number of the stars in the sky mattei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting CVS HEAD
Here's an excerpt from that page. Obviously, the hyperlinks are missing for some things, but I would reccommend rereading the page, specifically where it says, ...download a tarball of the released sources... These are release versions. If you want the CVS Head version, perhaps where it says, To check out code from our CVS repository: would be the place to look. Download Asterisk You can download a tarball of the released sources at ftp://ftp.asterisk.org/pub/asterisk. You can download the tarball files directly here: Asterisk Zaptel Libpri Asterisk-addons Asterisk-sounds You'll need Asterisk, and if you're using Digium's hardware you'll need zaptel. For T1 or E1 interfaces you'll also need libpri. You will need bison in order to build Asterisk. The ncurses and ncurses-devel packages are required if you wish to build the new tools (e.g. astman). Installation should be in this order: zaptel, libpri, Asterisk The fastest way to obtain Asterisk is to use CVSup. To check out Asterisk using CVSup, create a sup file as follows: *default host=cvs.digium.com *default base=/usr/src *default release=cvs tag=. *default delete use-rel-suffix asterisk libpri zaptel Perhaps call it asterisk-sup and put it in /usr/src Then simply: # cd /usr/src # cvsup asterisk-sup Or, you can obtain Asterisk by checking out a fresh copy from our CVS Server. To check out code from our CVS repository: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk On Apr 11, 2005 10:27 AM, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi, I want to download the CVS HEAD version. Any one can show how to get this version ? Is the version from: http://www.asterisk.org/index.php?menu=download the CVS Head version? How I can check if my version is CVS HEAD or not? Best Regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] timed Loop
This might seem really dumb but tack enough silence onto the back of your file to make it five minutes long. Then the message play for 5 minutes and repeats. Race The Tyrant Vanderdecken This was a dumb idea. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Monday, April 11, 2005 11:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] timed Loop I need to make a time loop in the Extensions.conf. I want it to play a file every 5 minutes on a call. If I can't use wait because it ignores all audio. Anyone have any suggestions? Regards, Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rebooting Asterisk box shows Asterisk failing to shutdown
hi, When I reboot my Fedora 3 box with Asterisk (latest version) I see Asterisk is failing to shutdown properly. All other processes shutdown and show success but Asterisk shows failed. What could be causing this. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to Reduce Latency
Folks, I have * running well but latency is too high (seems to be about 300-500 msec). This is on a lightly loaded Covad ADSL line running IAX to teliax, voipjet and voicepulse. Ping times to the teliax server are consistently in the 51-53 msec range. The others are similar. I am looking for someone who can tell me how to fix this. I am perfectly willing to buy hardware and to pay a reasonable amount for consulting services. If you can help, please contact me directly. Thanks, -- Art Z. -- Art Zemon, President Hen's Teeth Network http://www.hens-teeth.net/ Voice Fax: (866)HENS-NET or (636)447-3030 Customer Service Instant Messaging http://hens-teeth.net/chat.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - SS7 or ISDN
Brian Capouch wrote: Roy Sigurd Karlsbakk wrote: 1.Does Asterisk support SS7 and ISDN? ISDN is supported out of the box. SS7 support is (or will soon be?) supported by a commercial version of Asterisk. Search the list archives or post to asterisk-biz. Steve Underwood (here on the list) has a commercial ss7 solution for asterisk. Does anyone know how to find out any of the specs on the product, particularly what it costs to license? As reported by Steve on the list, the long waited SS7 is now finally available for Asterisk after several months of live beta testing with VoIP operators in Europe and Asia. Our SS7 LIBISUP solution is fully integrated with Asterisk over Zaptel and works with the Digium TE410/405/110 cards without any additional external equipment. I have just added some information in Wiki on our solution: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 The interested integrators can contact me off the list about the licensing and partnership. I have sent him a small river of mails over the past six or eight months asking that question, which seems pretty primal. I've never gotten a response. I apologize for my and Steve's part for the long silence and shutting the doors, while testing. I suppose now I'll hear that my mail agent is eating his responses. That would actually be *good* news. I hope we have now the really good news... Markku Cosini Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Low cost box for hosting Asterisk and at least one TDM400p
Hi, Can anyone recommend a very low cost box that could support Asterisk and at least one (preferably two) TDM400p cards and cost less that $150 (preferably under $100). The box should be able to run without a keyboard/mouse or CDROM. It also needs at least one Ethernet port. I know about converting the Linksys router to Asterisk but that does not give me the ability to connect POTS to the box. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax / realtime problems
Paul P. Pongco wrote: Hi Mat, It's Matthew :) I skip enabling pg and continue with make clean and make valgrind. That's fine. I found that out too. gdb backtrace still gives vague output: snip Still not clear, any pointers to make the backtrace more verbose? you probably need to recompile the mysql libraries with --enable-debug on the ./configure. Then recompile res_config_mysql. That should give us alot more backtrace stuff. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400p reliability????
Hi, I have been reading about some of the problems encountered with the TDM400p cards needing a reboot. I am still testing my system so I have not seen this yet. Also my card is showing as a version 2 when I run 'modprobe wtcdm' - is this problem in this card as well. If so can anyone recommend cards that can support FXS and FXO with PSTN (need standard POTS line connectivity) that are reliable and do not require this rebooting (I know about the CISCO cards and the 1760 but that solution is way too expensive ) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] append # to dial string
Thanks!
Right syntax - wrong box :-) (inter-iax between to *s - needed to apply
the suffix to the box talking directly TO the zap channel ... duhhh .)
Caught yet again by my own wrong assumtion
Eric Wieling wrote:
John Breeden wrote:
Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how
you can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? -
hex/octal whatever? I'm surprised that I can't find anything
searching the wiki or google.
Try something like this:
exten = _9XXX,1,Dial(Zap/4/${EXTEN}#)
Then you are doing something wrong. The above syntax is correct.
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