Re: [Asterisk-Users] 410P upgrade to 411P?
Yes, it can. I just had one of my old TE405Pv1 cards upgraded to a TE406P(same process as TE410P to TE411P upgrade). The cost is quoted at $895US. You do need to send it to Digium though, not sure if they have a partner in AUS that is able to do upgrades or not. Just contact digium and request a RMA for a firmware upgrade and an echo-can daughter-board install. MATT--- On 9/8/05, Rod Bacon <[EMAIL PROTECTED]> wrote: Does anyone know if the echo cancellation module can be retro-fitted to a 410Pto turn it into a 411P?--==Rod BaconEmpowered CommunicationsGround Floor, 102 York St. South Melbourne Victoria, Australia. 3205Phone: +613 99401600Fax: +613 99401650FWD: 512237 ICQ: 5662270==___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] 410P upgrade to 411P?
Does anyone know if the echo cancellation module can be retro-fitted to a 410P to turn it into a 411P? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.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] IAXy - no dailtone
On Thu, September 8, 2005 0:11, Jimmy said: > I have a brand new IAXy I'm playing with. I do not get a dialtone on > the phone, or any response at ll on the phone. No sound, no dialing, no > ringing. The phone and wire are tested and known to be good. I think I > have it setup correctly. When I give the iaxprov command I get this: > > #iaxyprov 192.168.1.90 iaxy.conf > 02: > c0 a8 01 5a That's 192.168.1.90, looks good! > 05: > 11 d9 > 03: > ff ff ff 00 That's 255.255.255.0, looks good too! > 04: > c0 a8 01 7d That's 192.168.1.125, still looks good! > 0d: > 00 00 00 04 > 0f: > c0 a8 01 c7 That's 192.168.1.199, still looks even better! > 10: > 11 d9 > 06: > 69 61 78 79 That's 'iaxy', good! > 07: > 70 61 73 73 77 6f 72 64 That's 'password', still good! > 0c: > 00 00 00 01 > Provisioning is 60 bytes > Total packet is 74 bytes > Got response back from '192.168.1.90' > Looks like the iaxy is configured conform the iaxy.conf info... > > Here is my iaxy.conf: > ; > ; IAXY Provisioning description > ; > ;dhcp > ip: 192.168.1.90 > netmask: 255.255.255.0 > gateway: 192.168.1.125 > codec: ulaw > ;codec: adpcm > server: 192.168.1.199 > ;altserver: 192.168.0.2 > user: iaxy > pass: password > register > ;heartbeat > ;debug > ; > ; Feature tuning (default is all enabled) > ; > ;disablecid > ;disablecw > ;disablecidcw > ;disable3way > > > The IP addresses here are all correct. > > Here's the relevant portion of iax.conf: > > [iaxy] > type=friend > user=iaxy > host=dynamic > secret=password > context=incoming > disallow=all > allow=ulaw > callerid="My IAXy" <(555) 555-1234> > trunk=no > > The BLUE light on the IAXy is lit. The ORANGE light blinks about once > every 7 seconds. I can dial this extension from another phone, and the > ORANGE light blinks rapidly while the phone should be ringing, but it > doesn't ring. And, as I stated earlier, the phone has no response at > all. No dialtone, no dialing, no ringing. > I'd almost suspect it is defective or lacking sufficient power... You *have* used the supplied PSU? > Have I missed something obvious? Is there some other test I can try? > Not that I know of... You could try dialling from the connected phone even though there is no dial tone... It could also be that the telephone you are using is defective or wired in a non-standard way... When the led blinks quickly, will you get a connection when you pick it up? What do the leds do when you pick up the phone? > Thanks in advance for any input. > > Jimmy Madden Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] I should never be called!
Hi, My configuration is pri - * - iaxclient based phone. Almost every day asterisk log file is filled with strange lines as follow. - Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! Sep 8 13:36:41 NOTICE[4339]: I should never be called! -- My * version is 1.0.7 running on tyan dual opteron board and 2 E1 sangoma card. iax.conf --- [general] port=5036 disallow=all tos=0x04 qualify=no [agent] type=friend username=agent secret=agent context=agent host=dynamic notransfer=yes callerid=20005000 zapata.conf --- [channels] context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national nationalprefix= signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes callprogressdetect=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=2.0 txgain=-4.0 group=1 channel => 1-15 channel => 17-31 channel => 32-46 channel => 48-62 -- Anyone please help? Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk, SIP, Re-INVITEs and different contexts
Hello Olle,
>It is easier to turn it around:
>Asterisk will issue a re-invite unless there is a reason
>to keep the audio going through Asterisk
>* NAT traversal issues
>* Canreinvite=no
>* Anything that needs asterisk to listen for DTMF in call
>* Codecs that needs to be transcoded
Ok, let's dig into this issue. Here is my test case.
Asterisk and two Cisco 7960 phones:
1. Extension 2780 with IP address 192.168.128.165 in context [internal]
2. Extension 1001 with IP address 192.168.128.172 in context [testing]
3. Asterisk is in the same subnet with IP 192.168.128.50
There is no NAT.
extensions.conf
[internal]
exten => 1001,1,Dial,local/[EMAIL PROTECTED]
exten => 2780,1,NoOp()
exten => 2780,2,Dial,SIP/2780|15
[testing]
exten => 2780,1,Dial,local/[EMAIL PROTECTED]
exten => 1001,1,NoOp()
exten => 1001,2,Dial,SIP/1001|15
sip.conf
[2780]
context=internal
type = friend
secret=2780
host = dynamic
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=ulaw
allow=g729
canreinvite=yes
[1001]
context=testing
type = friend
secret = 1001
host = dynamic
dtmfmode=rfc2833
qualify=2000
canreinvite=yes
disallow=all
allow=ulaw
allow=g729
Is there any questions with extensions.conf or sip.conf?
Ok, let move forward. Please see my comments to the end of this message.
Call from 2780 to 1001, this is short Ethereal trace:
---
Source Destination Protocol Info
192.168.128.165 192.168.128.50 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED], with session description
192.168.128.50 192.168.128.165 SIP Status: 407 Proxy
Authentication Required
192.168.128.165 192.168.128.50 SIP Request: ACK
sip:[EMAIL PROTECTED]
192.168.128.165 192.168.128.50 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED], with session description
192.168.128.50 192.168.128.165 SIP Status: 100 Trying
192.168.128.50 192.168.128.172 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED]:5060, with session description
192.168.128.172 192.168.128.50 SIP Status: 100 Trying
192.168.128.172 192.168.128.50 SIP Status: 180 Ringing
192.168.128.50 192.168.128.165 SIP Status: 180 Ringing
192.168.128.172 192.168.128.50 SIP/SDP Status: 200 OK, with session
description
192.168.128.50 192.168.128.172 SIP Request: ACK
sip:[EMAIL PROTECTED]:5060
192.168.128.50 192.168.128.165 SIP/SDP Status: 200 OK, with session
description
192.168.128.50 192.168.128.172 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED]:5060, with session description
192.168.128.165 192.168.128.50 SIP Request: ACK
sip:[EMAIL PROTECTED]:5060
192.168.128.50 192.168.128.165 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED]:5060, with session description
192.168.128.172 192.168.128.50 SIP/SDP Status: 200 OK, with session
description
192.168.128.50 192.168.128.172 SIP Request: ACK
sip:[EMAIL PROTECTED]:5060
192.168.128.165 192.168.128.50 SIP/SDP Status: 200 OK, with session
description
192.168.128.50 192.168.128.165 SIP Request: ACK
sip:[EMAIL PROTECTED]:5060
192.168.128.172 192.168.128.50 SIP Request: BYE
sip:[EMAIL PROTECTED]:5060
192.168.128.50 192.168.128.172 SIP Status: 200 OK
192.168.128.50 192.168.128.165 SIP/SDP Request: INVITE
sip:[EMAIL PROTECTED]:5060, with session description
192.168.128.165 192.168.128.50 SIP/SDP Status: 200 OK, with session
description
192.168.128.50 192.168.128.165 SIP Request: ACK
sip:[EMAIL PROTECTED]:5060
192.168.128.50 192.168.128.165 SIP Request: BYE
sip:[EMAIL PROTECTED]:5060
192.168.128.165 192.168.128.50 SIP Status: 200 OK
On Asterisk console:
-- Executing Dial("SIP/2780-1555", "local/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- Executing NoOp("Local/[EMAIL PROTECTED],2", "") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1001|15") in new stack
-- Called 1001
-- SIP/1001-667b is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/1001-667b answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 stopped sounds
-- Local/[EMAIL PROTECTED],1 answered SIP/2780-1555
-- Attempting native bridge of SIP/2780-1555 and SIP/1001-667b
*CLI> sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg
192.168.128.172 100175e28b2556a 00103/0 ulawTx: ACK
192.168.128.165 2780003094c2-bf 00102/00102 ulawTx: ACK
*CLI> show channels verbose
Channel Context Extension Prio State Application Data
CallerID Duration Accountcode BridgedTo
SIP/1001-667b testing 1 Up Bridged Call SIP/2780-1555
1001 SIP/2780-1555
SIP/2780-1555 testing 1001 2 Up Dial SIP/1001|15
2780 00:00:17 SIP/1001-667b
---
Now, call from 1001 to 2780:
Source Destination Protocol Info
192.168.128.172
[Asterisk-Users] OT: Differences between test equipment
Hello! Given the current discusison regarding ztmonitor, line testing, etc., I've been looking into purchasing a used transmission test set. From my research, it seems that there are two items that might fit the bill: the HP 3551A and the HP 4935A. I know nothing about these specific devices. I *do* have a good background in electronics, and I understand the concept what they're measuring and why., but I know nothing about the specifics of how this relates to transmission test sets! :) In fact, I'm not even sure that these are indeed the right devices for the job. Could someone who is familiar with either of these devices tell me if they will fit the bill? And if possible, which of these has more useful features as telephone line test equipment? I'm handy with an oscilloscope, function generator, VOM, etc. Will I be able to drive either of these? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.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] RDNIS
Your telco doesn't supply RDNIS (aka OCN) by default. I have two PRIs, each from a different provider. "Out-of-the-box", one PRI supports RNDIS/OCN and the other does not. I waiting for RDNIS/OCN to be enabled on the latter -- it was just a question of explaining what I wanted in the correct terms ie, explaining it's the OCN (orginial called number), isdn IE 115, that I wanted.On 9/7/05, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote: Anyone had any success using RDNIS? I have a number (not on our PRI) being forwarded to another number (on our pri) by the CLEC. When I call the first number, it goes to the number on our PRI and if I understand correctly the RDNIS should be populated with the first number. It's empty though. Any ideas? -Jonathan ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Sipura-2002 Can not make outgoing calls, incoming calls works OK
Sipura-2002 CAN NOT dial out, incoming call works OK. I just got a new Sipura-2002 to my collection (I have few Sipura-3000 units that work OK). I setup the unit, Sipura-2002 to register with Asterisk and it registered OK. The unit will accept the call but I can not make a call out. My sip.conf entry: [SPA-2] ; incoming/outgoing calls on FXS Sipura-2002-Line1 ext.711 type=friend secret=711 username=711 mailbox=711 host=dynamic port=5068 ; port on FXS line dtmfmode=rfc2833 nat=no context=incoming callgroup=1 pickupgroup=1 Dial Plan on Sipura-2002: (xx.|*xx.|#xx.) (this dial plan works OK on Sipura-3000) I tried to compare the setup of 2002 unit to 3000 but I can not find anything that would be blocking outgoing calls. The firmware on Sipura-2002: Software Version:3.1.5 When I try to make a call out the asterisk is not registering anything from the unit. I can not figure out why. -- #Joseph ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk frequently dead
Hi,
Sorry about the lack of information...
I use RHEL 4, asterisk cvs stable v1.0 and compile it myself..
It was worked well..
Asterisk was run stable in old platform (use duron), but then when I
upgraded it to P4, the problem is exists.
The weird things is I set asterisk in the same exact machine and the
problem only lies in this one..
The others run stable.
Maybe it's because I do share interrupt for asterisk? Will it help for
asterisk stability?
Here is output from lspci:
00:00.0 Host bridge: Silicon Integrated Systems [SiS]
661FX/M661FX/M661MX Host (rev 11)
00:01.0 PCI bridge: Silicon Integrated Systems [SiS]: Unknown device
0003
00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS964 [MuTIOL
Media IO] (rev 36)
00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev
01)
00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS]
Sound Controller (rev a0)
00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.2 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0
Controller
00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900
PCI Fast Ethernet (rev 90)
00:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
00:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
00:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
01:00.0 VGA compatible controller: Silicon Integrated Systems [SiS]
661FX/M661FX/M661MX/741/M741/760/M760 PCI/AGP
And here is output from cat /proc/interrupts
CPU0
0: 666453883 XT-PIC timer
1: 16 XT-PIC i8042
2: 0 XT-PIC cascade
5: 666283487 XT-PIC ohci_hcd, wctdm
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi, ehci_hcd
10: 666272466 XT-PIC SiS SI7012, ohci_hcd, wctdm
11: 697289929 XT-PIC ohci_hcd, wctdm, eth0
12: 66 XT-PIC i8042
14: 1557961 XT-PIC ide0
15: 1557358 XT-PIC ide1
NMI: 0
ERR: 0
Already do make clean, make and make install in the new platform.
Seems do not help at all...
* sigh *
Can you pinpoint what causes it to crash?
This is a tough question...I have no idea of what causing this or what should I do right now...
Perhaps somebody willing to give me 5 minutes tutor of using gdb?
I'm in process of learning it...Gotta be careful cause the system is used by more than 10 person (well, I'm getting tired of apologizing anyway :P)
Thanks,
Best Regards,
Stevanus
Andrew Kohlsmith wrote:
On Wednesday 07 September 2005 22:56, stevanus wrote:
My asterisk is frequently dead by itself.
It leaves messages:
/usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Someone new who's left us a wealth of information so we can diagnose the
problem quickly and help him find a timely solution.
When you take your car to the mechanic, do you simply say "It's broken. It
doesn't run the way it should." or do you give him some details. In this
case:
- Distribution of Linux
- Source of your Asterisk binaries (distribution packages, did you compile
yourself?)
- Version of Asterisk
- Has it ever worked
- Can you pinpoint what causes it to crash
I mean honestly, how do you expect us to help?
-A.
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[Asterisk-Users] Not can call to PSTN
Hello I have installed asterisk with a card X100P. receives calls but when doing the call to the PSTN. says that there are circuits no available... I have given to many returns but profit not to make work it I need that they help me with ideas by where to look for. Thanks (pardon my ingles little) -- Diego Ramirez D. ACODIN ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk frequently dead
On Wednesday 07 September 2005 22:56, stevanus wrote:
> My asterisk is frequently dead by itself.
>
> It leaves messages:
>
> /usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core
> dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} Asterisk ended with exit status 139
> Asterisk exited on signal 11.
> Automatically restarting Asterisk.
>
> Anyone has any idea of the cause?
Someone new who's left us a wealth of information so we can diagnose the
problem quickly and help him find a timely solution.
When you take your car to the mechanic, do you simply say "It's broken. It
doesn't run the way it should." or do you give him some details. In this
case:
- Distribution of Linux
- Source of your Asterisk binaries (distribution packages, did you compile
yourself?)
- Version of Asterisk
- Has it ever worked
- Can you pinpoint what causes it to crash
I mean honestly, how do you expect us to help?
-A.
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[Asterisk-Users] asterisk frequently dead
Hi,
My asterisk is frequently dead by itself.
It leaves messages:
/usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Anyone has any idea of the cause?
Thanks..
Best Regards,
Stevanus
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Re: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Hi,
I have similar problems like you.
In the past, I did adjusted my RX and TX gain, but didn't know if it has
been optimal yet.
Fxotune is seemed do not working, perhaps caused of my asterisk's
version ( I use stable v1.0)..
Just curious, is rx and tx gain really a sole setting option here in
order to make things the way it's meant to be? Or is there others?
FYI, my tdm04b occasionally don't detect call-in as well as hangup signal.
I've searched in the wiki and have activated hanguponpolarity swicth.
But I don't notice any difference at all.
Any help would be greatly appreciated. (I've asked this in another
thread, but got no respon :( )
Best Regards,
Stevanus
canuck15 wrote:
This may or may not be related but have you tried adjusting your RX and TX
gains? I see both are at the default (0.0) which leads me to believe you
have not. Search the Asterisk Wiki for the procedure.
-Original Message-
From: Faris Raouf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 07, 2005 12:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone calls in, and then hangs up at any time after it starts ringing,
Asterisk carries on as though the caller never hung up.
I've tried raising BATT_THRESH to 8 in wcfxs.c and re-compiling zaptel (this
was the only thing that Google came up with to help me, although others do
seems to have had similar problems to mine at various times), but it has
made no difference at all.
The second problem is that Hangup does not hangup. The channel stays open
until I stop asterisk.
Note: When MAKING a call on the FXO, when I terminate the call on my SIP
phone the line does drop correctly. The problem appears to be related to
incoming calls only.
I'm based in the UK. Using RedHat 9 with zaptel-1.0.9.1, asterisk-1.0.9 (and
chan_capi-0.5.4)
Thanks in advance for any ideas.
Faris.
*
Here's my initialisation script:
modprobe zaptel
modprobe wctdm opermode=UK
/sbin/ztcfg -
capiinit
safe_asterisk
zapata.conf
[trunkgroups]
; nothing in here
[channels]
rxwink=300 ; (I tried commenting this out. Make no difference)
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=no
sendcalleridafter=1
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
progzone=uk
; module 0 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 1
; module 1 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 2
; module 2 on card is an FXO
signalling=fxs_ks
language=en
context=faris
channel => 3
zaptel.conf
fxoks=1-2
fxsks=3
loadzone=uk
defaultzone=uk
and in extensions.conf
[faris]
exten => s,1,NoOp(cid=${CALLERID})
exten => s,2,Wait(10)
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Playback(some-long-message) exten => s,6,Hangup
The long wait(10) is just there to see what happens. Removing it makes no
difference. Basically whenever a call comes in, no matter when the caller
hangs up, Asterisk continues with the call to the end (i.e. plays long
message).
What's more, the Hangup at the end has no effect. The line is not dropped.
The line is not ever dropped in fact.
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Re: [Asterisk-Users] Business telephones
On Tuesday 06 September 2005 09:12, Steve Langstaff wrote: > Are you starting from scratch, or do you already have a phone system that > does some/all of this? I have the digium dev kit (xp100 PSTN & TDM400 w/4 fxs modules) - I use pstn on the xp100 and 4 (well 3, as one fxs is dead) single line phones. I'm looking for business type sip phones to expand the test system. I have used a number of business systems in the past. Hitachi, toshiba, avaya, norstar/meridian. Personally, I like Iwatsu adix. http://www.iwatsu.com/Products/EnhancedPhone.html Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM400P problems
On Friday 02 September 2005 11:38, Andrew Kohlsmith wrote: > On Friday 02 September 2005 12:57, Martin wrote: > > I would be really curious if anyone else has the capability (digital > > multimeter) and enough knowledge to test their incoming CO (Central > > Office line) and board outputs. > > I will test mine at lunchtime today. > > -A. What was the result ??? ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Re: ztcfg Kills My Dial Tone [solved]
On Wed, Sep 07, 2005 at 11:50:28AM -0500, Shaw Terwilliger wrote: > Right after I reboot, and modprobe wct4xxp, my analog phone connected > to port 13 of the first channel bank (first FXS port) gets a dial tone. > Asterisk is not running yet, and I have NOT run ztcfg. I've solved my own problem, with a little help from Rhino (and some poking around). I think my problem was that an older version of the wct4xxp module was sitting around in the kernel's module dirctory, so I had an inconsistency between the userspace tools and the kernel drivers. Also, I had my signalling misconfigured the first time I brought up the channel banks, and even though I had corrected this in zaptel.conf, I had to tell the channel banks to "Auto T1" themselves. Now I get nice, strong dial tones on my FXS interfaces, and my FXO interfaces happily listen to POTS lines. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Hack for Canadian weather
[your-context] include => app-canadian-weather [app-canadian-weather] exten => *55,1,Answer() exten => *55,2,Playback(pls-wait-connect-call) exten => *55,3,System(/etc/asterisk/weather.sh | text2wave -o /var/lib/asterisk/sounds/weather.ulaw -otype ulaw -) exten => *55,4,Playback(weather) /etc/asterisk/weather.sh: rm -f /var/lib/asterisk/sounds/weather.ulaw rm -f /etc/asterisk/textforecast_e.html?Bulletin=fcpn16.cwwg wget -q http://weatheroffice.ec.gc.ca/forecast/textforecast_e.html?Bulletin=fcpn16.c wwg grep -A5 "City of Edmonton" textforecast_e.html?Bulletin=fcpn16.cwwg ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Problem with PRI channels, restarted after every call.
Hi,
I got a problem with PRI that I’m not sure how to
solve.
Asterisk sits between PABX and PRI.
PRI is span 1 and PABX is span 2.
After every single call (no matter in what direction) I
get “pri_fixup_principle: Call specified, but not found?” and “pri_dchannel:
Hangup on bad channel” messages and the channel in question is restarted.
As far as I can see, all calls complete fine.
What could cause this? I’d appreciate any help given.
Here is the debug log.
-- Goto (PRI_NET_Out,96422241,1)
-- Executing Dial("Zap/92-1",
"Zap/r0/96422241|300|j") in new stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called r0/96422241
-- Zap/15-1 is proceeding passing it to
Zap/92-1
-- Zap/15-1 is ringing
-- Zap/14-1 is ringing
-- Zap/14-1 answered Zap/93-1
-- Attempting native bridge of Zap/93-1
and Zap/14-1
-- Zap/15-1 answered Zap/92-1
-- Attempting native bridge of Zap/92-1
and Zap/15-1
-- Channel 0/31, span 3 got hangup
request
-- Hungup 'Zap/14-1'
== Spawn extension (PRI_NET_Out, 97354333, 1) exited
non-zero on 'Zap/93-1'
-- Hungup 'Zap/93-1'
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:03 WARNING[13375]: chan_zap.c:8701 pri_dchannel: Hangup on bad channel
0/14 on span 1
Sep 8 10:36:07 WARNING[13375]: chan_zap.c:7651
pri_fixup_principle: Call specified, but not found?
Sep 8 10:36:07 WARNING[13375]: chan_zap.c:8701
pri_dchannel: Hangup on bad channel 0/14 on span 1
-- B-channel 0/14 restarted on span 1
Boris Bakchiev
Jildent Pty Ltd
Tel: + 61 3 8080 5898
Fax: +61 3 9811 4716
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RE: [Asterisk-Users] Motherboard and processor recommendations
Regarding Chipsets/Motherboards. I would stay FAR away from cheap ones. Any chipset/motherboard that electrically and logically separates some PCI slots (ie. interrupts) from onboard peripherals (network controller, VGA, USB etc.) makes compatibility issues with Digium cards much less likely. Many of the newer Intel chipsets do this. The Xeon chipsets/motherboards are the best IMHO because they usually have PCI-X slots connected directly to the memory controller hub, that you can put your Digium card(s) in, which are completely separate from the peripherals and PCI slots on the I/O controller hub. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 07, 2005 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Motherboard and processor recommendations On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote: > Hi All, > > For sometime now I've been searching the wiki and googling, but I > think I'm missing some of the very important answers. So I'll have to > ask this to the list. > > I'm trying to decide on the right motherboard and processor. Here are > my > questions: > > 1. Would I have problems with all-onboard motherboards (Onboard VGA, > LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard > VGA on wiki. Considering the exceptional quality of graphics you'll need with Asterisk, and VGA-compatible adapter would suffice. The on-board one would be more than enough. Ditto for the sound card, at least in most cases. As for the network adapter: Are you going to get anything close to saturating the card? I figure that the efficiency of the network adapter and its driver will not be your bottleneck. Most of the WAN-oriented systems would have worked fine with an old 10Mbps card, probably without a noticable performance hit (right?). So their quality is not much of an issue. If you have the extra space, you can always add an extra one in an expansion slot. But it should not be required. An extra raid controller is something you may consider. But then-again, if it is a cheap software-based raid, it is practically the same as using linux for that (but with more problematic drivers). But it is for you to decide if it is worth the extra cost. > > 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old > SiS chipset problem on wiki. There is much voodoo about this. There are good and bad boards made with each of those chipsets. In fact, for practically each model of board that has been sold for over a month or so, you'll probably find someone in this list who had bad experience with it. > > 3. Which processor has the least support problems: P4 (478 or LGA775, > or even EMT64) or AMD64 ? For example, in G729 config file Athlon > comment reads as "untested" (so far I don't have problems), and there > is no config option for AMD64 at all. There is no mention of EMT64 > either. Is anything processor dependant in codecs/transcoding, echo > cancellation, busy detect and similar software, i.e. in dsp routines in general ? First of all, what do you intend to do? Much transcoding? How many lines? Because if the load on the CPU will be light enough, than the CPU brand won't really matter, you know. > I think this is a very complicated issue, and given so many variables > perhaps luck plays an important part. I figure some people on this list will happily sell you pre-configured systems. Or at least pre-built ones. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Need Help - Losing first few seconds of call when using Broadvoice
Hi, I am fairly new at Asterisk, but have been using Asterisk at Home for a few months now. I have 3 Polycom IP600 phones running with Broadvoice and Voipjet services. When I place calls to the PSTN, I am losing the first few seconds of the call. This is especially true with the SIP client, Broadvoice. I have tested it by calling my fax machine. It seems that when the fax machine rings, I am hearing nothing on my Polycom, but when it connects on the fax is when I here a ring. This is the length of the delay. Usually I miss the greeting of the party I have called when using Broadvoice. When I use the IAX client, Voipjet, the delay is fairly negligible. I am using a DLink DGL-4300 router, which has been advertised as a "Gaming Router". It offers an upstream QOS, as well as port forwarding. I have used and removed both for testing purposes. Any help at all will be not only appreciated, but will help my very tired eyes after searching the net for any answers to previous questions of this type. ___ --Bandwidth and Colocation sponsored by Easynews.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] Motherboard and processor recommendations
On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote: > Hi All, > > For sometime now I've been searching the wiki and googling, but I think I'm > missing some of the very important answers. So I'll have to ask this to the > list. > > I'm trying to decide on the right motherboard and processor. Here are my > questions: > > 1. Would I have problems with all-onboard motherboards (Onboard VGA, > LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard VGA > on wiki. Considering the exceptional quality of graphics you'll need with Asterisk, and VGA-compatible adapter would suffice. The on-board one would be more than enough. Ditto for the sound card, at least in most cases. As for the network adapter: Are you going to get anything close to saturating the card? I figure that the efficiency of the network adapter and its driver will not be your bottleneck. Most of the WAN-oriented systems would have worked fine with an old 10Mbps card, probably without a noticable performance hit (right?). So their quality is not much of an issue. If you have the extra space, you can always add an extra one in an expansion slot. But it should not be required. An extra raid controller is something you may consider. But then-again, if it is a cheap software-based raid, it is practically the same as using linux for that (but with more problematic drivers). But it is for you to decide if it is worth the extra cost. > > 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old SiS > chipset problem on wiki. There is much voodoo about this. There are good and bad boards made with each of those chipsets. In fact, for practically each model of board that has been sold for over a month or so, you'll probably find someone in this list who had bad experience with it. > > 3. Which processor has the least support problems: P4 (478 or LGA775, or > even EMT64) or AMD64 ? For example, in G729 config file Athlon comment > reads as "untested" (so far I don't have problems), and there is no config > option for AMD64 at all. There is no mention of EMT64 either. Is anything > processor dependant in codecs/transcoding, echo cancellation, busy detect > and similar software, i.e. in dsp routines in general ? First of all, what do you intend to do? Much transcoding? How many lines? Because if the load on the CPU will be light enough, than the CPU brand won't really matter, you know. > I think this is a very complicated issue, and given so many variables > perhaps luck plays an important part. I figure some people on this list will happily sell you pre-configured systems. Or at least pre-built ones. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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] Want to use a remotely location POTS phone
For ONE line ? I'd try a sipura spa3k. In location A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Campbell Sent: Thursday, September 08, 2005 02:18 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Want to use a remotely location POTS phone I have 2 locations connection via high speed internet. At location B, I have a regular old phone line I want to be able to use seemlessly from location A. What would be required to do this both software and hardware wise? I (think I) understand a SIP adapter at location A, and a server running * at location B, but what type of hardware will let me connect the POTS line to the * server? Thanks! -- Patrick Campbell ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Want to use a remotely location POTS phone
On Wednesday 07 September 2005 19:18, Patrick Campbell wrote: > I have 2 locations connection via high speed internet. At location B, I > have a regular old phone line I want to be able to use seemlessly from > location A. What would be required to do this both software and hardware > wise? I (think I) understand a SIP adapter at location A, and a server > running * at location B, but what type of hardware will let me connect the > POTS line to the * server? Any old FXO port or just simply call forward the line somewhere local (if it's local) -A. ___ --Bandwidth and Colocation sponsored by Easynews.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] Several SIP clients behind router register withthe same IP, messing up call routing, any ideas?
Title: Message Yes, NAT=yes in all users and "nat=no" is commented out in sip.conf Is there any other place to check? Thanks, Roman -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FlobiSent: Mittwoch, 7. September 2005 22:48To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Several SIP clients behind router register withthe same IP, messing up call routing, any ideas? Do you have NAT turned on? On 9/7/05, Roman Zhovtulya <[EMAIL PROTECTED]> wrote: Dear all,Has anyone seen this before and can suggest a solution?I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address.Now, wenn all are registered, all the calls get routed to the client thatregistered most recently, but not to the correct client.Also, there seems to be some problems registering all 3 clients simultaneously.Could anyone help, please?Thank you very much,Roman Zhovtulya___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Automated Signature: This message is from Flobi of Flobi.com.Visit my website if you like: http://www.flobi.com/Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Want to use a remotely location POTS phone
I have 2 locations connection via high speed internet. At location B, I have a regular old phone line I want to be able to use seemlessly from location A. What would be required to do this both software and hardware wise? I (think I) understand a SIP adapter at location A, and a server running * at location B, but what type of hardware will let me connect the POTS line to the * server? Thanks! -- Patrick Campbell ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] RDNIS
Title: Message Anyone had any success using RDNIS? I have a number (not on our PRI) being forwarded to another number (on our pri) by the CLEC. When I call the first number, it goes to the number on our PRI and if I understand correctly the RDNIS should be populated with the first number. It's empty though. Any ideas? -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] te110p
hi all. any body have the zaptel.conf and zapata.conf files for the card te110p ? please another question when i start asterisk the te110p give the al channels busy is that correct? [EMAIL PROTECTED] root]# cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4 RED 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) ___ --Bandwidth and Colocation sponsored by Easynews.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] IVR Documentation and Samples.
Juan Salas, Are you use IVR System with database? Case yes, please send me one sample. Regards. PJSantos.Juan Salas <[EMAIL PROTECTED]> escreveu: Hi Evrybody. I have tested Asterisk Real Time (asterisk version 1.2 beta) with postgresql (and unixODBC) and it works very well! Now I will try with Oracle, Have somebody work with it? Regards Jsalas. -Mensaje original-De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:31 PMPara: Asterisk-Users@lists.digium.comAsunto: [Asterisk-Users] IVR Documentation an Sample. Hi everybody. I need documentation and sample, about IVR. Sample about database access with IVR+Asterisk, if its possible. Thanks. Paulo Santos Brasil/RJ. __Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___--Bandwidth and Colocation sponsored by Easynews.com --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__Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___ --Bandwidth and Colocation sponsored by Easynews.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] g729 test
On Wednesday 07 September 2005 18:32, Michael Welter wrote: > Is there a method to test, prior to initiating a call, whether a g729 > codec is available? Will ChanIsAvail test g729 availability? Only if you do your own refcounting with a DB variable, I think. It's on my personal wishlist. -A. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users]
Hi Evrybody. I have tested Asterisk Real Time (asterisk version 1.2 beta) with postgresql (and unixODBC) and it works very well! Now I will try with Oracle, Have somebody work with it? Regards Jsalas. -Mensaje original-De: PJ Santos [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, September 07, 2005 6:31 PMPara: Asterisk-Users@lists.digium.comAsunto: [Asterisk-Users] IVR Documentation an Sample. Hi everybody. I need documentation and sample, about IVR. Sample about database access with IVR+Asterisk, if its possible. Thanks. Paulo Santos Brasil/RJ. __Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] g729 test
My preferred LD vendor requires g729 and SIP. Is there a method to test, prior to initiating a call, whether a g729 codec is available? Will ChanIsAvail test g729 availability? ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] IVR Documentation an Sample.
Hi everybody. I need documentation and sample, about IVR. Sample about database access with IVR+Asterisk, if its possible. Thanks. Paulo Santos Brasil/RJ.__Converse com seus amigos em tempo real com o Yahoo! Messenger http://br.download.yahoo.com/messenger/ ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Remote Provisioning for the PA1688 phones.
Dear helpers! I am looking for the software and information about remote provisioning for the PA1688 phones to upload configuration files using TFTP, FTP, or HTTP. Do you guys know how and where I could find those information! I would be very appreciated for your help! Regards, Ian. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] IAXy - no dailtone
I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04: c0 a8 01 7d 0d: 00 00 00 04 0f: c0 a8 01 c7 10: 11 d9 06: 69 61 78 79 07: 70 61 73 73 77 6f 72 64 0c: 00 00 00 01 Provisioning is 60 bytes Total packet is 74 bytes Got response back from '192.168.1.90' Here is my iaxy.conf: ; ; IAXY Provisioning description ; ;dhcp ip: 192.168.1.90 netmask: 255.255.255.0 gateway: 192.168.1.125 codec: ulaw ;codec: adpcm server: 192.168.1.199 ;altserver: 192.168.0.2 user: iaxy pass: password register ;heartbeat ;debug ; ; Feature tuning (default is all enabled) ; ;disablecid ;disablecw ;disablecidcw ;disable3way The IP addresses here are all correct. Here's the relevant portion of iax.conf: [iaxy] type=friend user=iaxy host=dynamic secret=password context=incoming disallow=all allow=ulaw callerid="My IAXy" <(555) 555-1234> trunk=no The BLUE light on the IAXy is lit. The ORANGE light blinks about once every 7 seconds. I can dial this extension from another phone, and the ORANGE light blinks rapidly while the phone should be ringing, but it doesn't ring. And, as I stated earlier, the phone has no response at all. No dialtone, no dialing, no ringing. Have I missed something obvious? Is there some other test I can try? Thanks in advance for any input. Jimmy Madden ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] asterisk-statv2 showing blank screens
I have typed in the address the line you gave me. http://myserver/asterisk-stat/graph_stat.php?min_call=&fromstatsday_sday=01&days_compare=2&fromstatsmonth_sday=2005-05 came up blank screen again. No errors. and even started again with mysql 3.23, php, fedora 3, asterisk, asterisk mysql addon, oh323 driver. I can see the calls in the sql database. With a fresh install I only had two calls from sip to sip which I quickly did. < php info > --with-ttf isn’t in the list but everything else is and gd is installed. Should I be using the database layout from voip.org? Thank you, William Bohannan GRANT INSERT ON cdr.* TO [EMAIL PROTECTED] IDENTIFIED BY '8anana?!'; USE cdr; CREATE TABLE cdr ( uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '', accountcode varchar(20) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', clid varchar(80) NOT NULL default '', channel varchar(80) NOT NULL default '', dstchannel varchar(80) NOT NULL default '', lastapp varchar(80) NOT NULL default '', lastdata varchar(80) NOT NULL default '', calldate datetime NOT NULL default '-00-00 00:00:00', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags int(11) NOT NULL default '0' ); ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM400P not detecting hangup and not hanging up.
This may or may not be related but have you tried adjusting your RX and TX
gains? I see both are at the default (0.0) which leads me to believe you
have not. Search the Asterisk Wiki for the procedure.
-Original Message-
From: Faris Raouf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 07, 2005 12:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone calls in, and then hangs up at any time after it starts ringing,
Asterisk carries on as though the caller never hung up.
I've tried raising BATT_THRESH to 8 in wcfxs.c and re-compiling zaptel (this
was the only thing that Google came up with to help me, although others do
seems to have had similar problems to mine at various times), but it has
made no difference at all.
The second problem is that Hangup does not hangup. The channel stays open
until I stop asterisk.
Note: When MAKING a call on the FXO, when I terminate the call on my SIP
phone the line does drop correctly. The problem appears to be related to
incoming calls only.
I'm based in the UK. Using RedHat 9 with zaptel-1.0.9.1, asterisk-1.0.9 (and
chan_capi-0.5.4)
Thanks in advance for any ideas.
Faris.
*
Here's my initialisation script:
modprobe zaptel
modprobe wctdm opermode=UK
/sbin/ztcfg -
capiinit
safe_asterisk
zapata.conf
[trunkgroups]
; nothing in here
[channels]
rxwink=300 ; (I tried commenting this out. Make no difference)
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=no
sendcalleridafter=1
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
progzone=uk
; module 0 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 1
; module 1 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 2
; module 2 on card is an FXO
signalling=fxs_ks
language=en
context=faris
channel => 3
zaptel.conf
fxoks=1-2
fxsks=3
loadzone=uk
defaultzone=uk
and in extensions.conf
[faris]
exten => s,1,NoOp(cid=${CALLERID})
exten => s,2,Wait(10)
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Playback(some-long-message) exten => s,6,Hangup
The long wait(10) is just there to see what happens. Removing it makes no
difference. Basically whenever a call comes in, no matter when the caller
hangs up, Asterisk continues with the call to the end (i.e. plays long
message).
What's more, the Hangup at the end has no effect. The line is not dropped.
The line is not ever dropped in fact.
___
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RE: [Asterisk-Users] Desincripcion de la lista de Asterisk
Primero que nada, no les envies e-mails en castellano, ya que no lo van a entender. Segundo, envia en e-mail a [EMAIL PROTECTED] poniendo como Subject: Stop my subscription!!! Eso es todo. Saludos, Carlos From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Will Velez Sent: Wednesday, September 07, 2005 8:22 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Desincripcion de la lista de Asterisk Buenos días quiero que ya no me llegue mas correo electrónico de la lista Asterisk, como puedo hacerlo Gracias ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM04B + Voicemail poor Quality
[EMAIL PROTECTED] wrote on 07/18/2005 11:56:06 AM: > > > Recently, I installed TDM04B 4 FXO card on to my Asterisk box and > > installation went perfect. > > > > The only problem I am facing is the Voice mail has very poor quality > > when any users leave voice message via PSTN line. > > > > We can not hear either from the extension nor from the WAV email > > attached. > > > > Has anyone experienced this problem before, please help? > > Yes, its well known. See bug #2023 in the bug tracker. > > Kevin would like to address this along with addressing missed frames > (common complaint when attempting to use spandsp) before the next > formal Stable release. But, its too early to guess whether that > will actually happen (as of right now). Does this relate to the e-mail you just sent to the list regarding volume issues with voicemail files not saved as uncompressed WAV's? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] externpass in voicemail
Guys. Since ARA is not available in stable 1.0.9 I was wondering how to use externpass in voicemail.conf to update mysql based voicemail password. My question is, can externpass send parameters to the called file? I couldnt find any examples on the wiki so, does anybody have any tips on this? Thx ! ___ --Bandwidth and Colocation sponsored by Easynews.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] Utility to find length of wav49 file
On 9/7/2005, "Malcolm Taylor" <[EMAIL PROTECTED]> wrote: >Thanks Flynn. > > Unfortunately the files aren't written by the voicemail application. > I was hoping that there was some little command-line utility which > would return basic sound information when passed the filename. > > Malcolm You could always use sox... I just ran it here $>sox -e stat Brett ___ --Bandwidth and Colocation sponsored by Easynews.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] Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
Do you have NAT turned on? On 9/7/05, Roman Zhovtulya <[EMAIL PROTECTED]> wrote: Dear all,Has anyone seen this before and can suggest a solution?I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address.Now, wenn all are registered, all the calls get routed to the client thatregistered most recently, but not to the correct client.Also, there seems to be some problems registering all 3 clients simultaneously.Could anyone help, please?Thank you very much,Roman Zhovtulya___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Automated Signature: This message is from Flobi of Flobi.com.Visit my website if you like: http://www.flobi.com/Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could anyone help, please? Thank you very much, Roman Zhovtulya ___ --Bandwidth and Colocation sponsored by Easynews.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] Speaking of Polycom phones...updated ROM: ouch!
At 12:38 9/7/2005 -0700, Jesse Keating wrote: On Wed, 2005-09-07 at 14:18 -0500, Doug wrote: > > I again followed instructions here: > http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501 So yeah, the instructions are a bit misleading. I had to set register to yes prior to the line information stuff. Without that the phone wouldn't register. Now it registers, and I still get 3 buttons dedicated to a single extension. Hey Jesse, I've tried just about every combo there is "yes", "Yes", "no", "No", "1", "0"--still no registration, phone icon is not solid black. Do you have any other ideas, or can you send me your HTML config pages? ___ --Bandwidth and Colocation sponsored by Easynews.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] Hosted PBX (vPBX) and Call/PickUP Groups
Why don't you try this: http://pbxfreeware.com/app_intercept.c On 9/7/05, René Mayorga <[EMAIL PROTECTED]> wrote: > Hi > Can you give me any hint on with file of the source you modify that > Value??? > > tnx > > On Wed, 2005-09-07 at 08:18 -0400, Flobi wrote: > > I'm not sure about why, but it's it is hardcoded into asterisk. Back > > when it was a limit of 31, I searched around and increased the value > > on my box and recompiled. It did not seem to adversely affect the > > system. > > > > On 9/7/05, René Mayorga <[EMAIL PROTECTED]> wrote: > > > Hi, > > > I'm working with this issue for a while, Now I already solve the > > > dialplan issues, but I still have a question about the Callgroups, > > > I read at www.voip-info.org that , there is a 63 limit of callgroups. > > > And I'm wondering why?? and if the 1.2.0beta version supported more than > > > 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any > > > unoficial patch for that ? > > > > > > Thanks in advance. > > > > > > -- > > > René Mayorga <[EMAIL PROTECTED]> > > > El Salvador Telecom S.A. de C.V. > > > > > > ___ > > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > > > > -- > René Mayorga <[EMAIL PROTECTED]> > El Salvador Telecom S.A. de C.V. > > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Asterisk with Vonage problems
Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's but Xten Eyebeam handles it correctly at least. I have tried pedantic=yes but no difference. Here is the sip.conf and the BYE dialog with numbers replaced: [vonage] type=friend secret= username= host=sphone.vopr.vonage.net dtmfmode=rfc2833 port=5061 fromuser= fromdomain=sphone.vopr.vonage.net canreinvite=no context=context insecure=very BYE sip:(PSTN Number)@216.115.20.171:5060 SIP/2.0 Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d Route: 216.115.20.171:5060> From: "Adrian" sphone.vopr.vonage.net>;tag=as74d54cec To: sphone.vopr.vonage.net:5061>;tag=2067764114 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 104 BYE Proxy-Authorization: Digest username="(Vonage No)", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="18861432149", response="5de1aaac0fa9db87sdfb074a1fe324b", opaque="" Content-Length: 0 --- == Spawn extension (default, 8(PSTN Number), 3) exited non-zero on 'SIP/370-29aa' Destroying call '[EMAIL PROTECTED]' <-- SIP read from 216.115.25.198:5061: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d From: "Adrian" sphone.vopr.vonage.net>;tag=as74d54cec To: sphone.vopr.vonage.net:5061>;tag=2067764114 Call-ID: [EMAIL PROTECTED] CSeq: 104 BYE Max-Forwards: 15 Content-Length: 0 The issue I think is that Asterisk uses the "To" tag from the 183 Session Progress instead of the tag from the 200 OK that Vonage sends. If anyone uses Vonage with Asterisk and it works fine for you (ie. landline hangs up when you hang up), can you please let me know which version you're using? (I'm using CVS HEAD from a couple of months ago and would like to know if an upgrade may fix the issue.) ___ --Bandwidth and Colocation sponsored by Easynews.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] Hosted PBX (vPBX) and Call/PickUP Groups
Hi Can you give me any hint on with file of the source you modify that Value??? tnx On Wed, 2005-09-07 at 08:18 -0400, Flobi wrote: > I'm not sure about why, but it's it is hardcoded into asterisk. Back > when it was a limit of 31, I searched around and increased the value > on my box and recompiled. It did not seem to adversely affect the > system. > > On 9/7/05, René Mayorga <[EMAIL PROTECTED]> wrote: > > Hi, > > I'm working with this issue for a while, Now I already solve the > > dialplan issues, but I still have a question about the Callgroups, > > I read at www.voip-info.org that , there is a 63 limit of callgroups. > > And I'm wondering why?? and if the 1.2.0beta version supported more than > > 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any > > unoficial patch for that ? > > > > Thanks in advance. > > > > -- > > René Mayorga <[EMAIL PROTECTED]> > > El Salvador Telecom S.A. de C.V. > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > -- René Mayorga <[EMAIL PROTECTED]> El Salvador Telecom S.A. de C.V. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Motherboard and processor recommendations
Hi All, For sometime now I've been searching the wiki and googling, but I think I'm missing some of the very important answers. So I'll have to ask this to the list. I'm trying to decide on the right motherboard and processor. Here are my questions: 1. Would I have problems with all-onboard motherboards (Onboard VGA, LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard VGA on wiki. 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old SiS chipset problem on wiki. 3. Which processor has the least support problems: P4 (478 or LGA775, or even EMT64) or AMD64 ? For example, in G729 config file Athlon comment reads as "untested" (so far I don't have problems), and there is no config option for AMD64 at all. There is no mention of EMT64 either. Is anything processor dependant in codecs/transcoding, echo cancellation, busy detect and similar software, i.e. in dsp routines in general ? 4. How important is the number of PCI slots? I mean, considering that I've read some comments on this list, which do not recommend more than 2 TDM cards on a single system (right?), 2-3 PCI slots should be enough, is this correct? (But beware this also means an all-onboard motherboard, in most cases.) I think this is a very complicated issue, and given so many variables perhaps luck plays an important part. I'd like to hear your experiences. Any links I wasn't able find are welcome too. Thanks, Soner ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone calls in, and then hangs up at any time after it starts ringing,
Asterisk carries on as though the caller never hung up.
I've tried raising BATT_THRESH to 8 in wcfxs.c and re-compiling zaptel (this
was the only thing that Google came up with to help me, although others do
seems to have had similar problems to mine at various times), but it has
made no difference at all.
The second problem is that Hangup does not hangup. The channel stays open
until I stop asterisk.
Note: When MAKING a call on the FXO, when I terminate the call on my SIP
phone the line does drop correctly. The problem appears to be related to
incoming calls only.
I'm based in the UK. Using RedHat 9 with zaptel-1.0.9.1, asterisk-1.0.9 (and
chan_capi-0.5.4)
Thanks in advance for any ideas.
Faris.
*
Here's my initialisation script:
modprobe zaptel
modprobe wctdm opermode=UK
/sbin/ztcfg -
capiinit
safe_asterisk
zapata.conf
[trunkgroups]
; nothing in here
[channels]
rxwink=300 ; (I tried commenting this out. Make no difference)
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=no
sendcalleridafter=1
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
progzone=uk
; module 0 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 1
; module 1 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 2
; module 2 on card is an FXO
signalling=fxs_ks
language=en
context=faris
channel => 3
zaptel.conf
fxoks=1-2
fxsks=3
loadzone=uk
defaultzone=uk
and in extensions.conf
[faris]
exten => s,1,NoOp(cid=${CALLERID})
exten => s,2,Wait(10)
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Playback(some-long-message)
exten => s,6,Hangup
The long wait(10) is just there to see what happens. Removing it makes no
difference. Basically whenever a call comes in, no matter when the caller
hangs up, Asterisk continues with the call to the end (i.e. plays long
message).
What's more, the Hangup at the end has no effect. The line is not dropped.
The line is not ever dropped in fact.
___
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RE: [Asterisk-Users] Which Linux distribution?
I can't understand why anyone would use Fedora Core. Sure it 'can be' quite stable depending on what your doing but it is not considered a production ready OS. Any of the Red Hat Enterprise edition clones such as CentOS or White Box Enterprise Linux are a MUCH better alternative IMHO. I don't have any direct experience with CAPI so I can comment on that specifically. -Original Message- From: YT Lim [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 06, 2005 9:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which Linux distribution? We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea Do you Yahoo!? The New Yahoo! Movies: Check out the Latest Trailers, Premiere Photos and full Actor Database. http://au.movies.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.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] "-- PROGRESS with cause code 34 received"?
On Wed, Sep 07, 2005 at 01:47:49PM +0200, Roy Sigurd Karlsbakk wrote: > hi > > i get these messages every now and then > > "-- PROGRESS with cause code 34 received" > > wtf is this? Nothing to see here, move along :-) Seriously though, it's basically just and interesting message to see. The cause code IE withing the progress message was set to 34 (You can look up what that means in the Q.931 spec). -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.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] Speaking of Polycom phones...updated ROM: ouch!
On Wed, 2005-09-07 at 14:18 -0500, Doug wrote: > > I again followed instructions here: > http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501 So yeah, the instructions are a bit misleading. I had to set register to yes prior to the line information stuff. Without that the phone wouldn't register. Now it registers, and I still get 3 buttons dedicated to a single extension. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM card and voicemail volume
> [EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM: > > > I was able to raise the volume from inaudible to acceptable by > > increasing the RxGain in zapata.conf by 5db. I'd rather not go the > > uncomressed wav route, as it will chew up storage in my email system. > > I know I'm way behind on reading this, but I thought I would follow up. > > According to this message: > > > http://lists.digium.com/pipermail/asterisk-users/2004-November/072990.html > > the reason that uncompressed WAV files are louder is that the software > that saves the WAV file is amplifying the volume of the files by shifting > the data two bits to the left (or making it 4x louder). It is in no way > "fixing" the underlying problem of the file being too quiet; it is just > throwing away dynamic range in order to amplify the file. > > Now that may not be a bad solution: if you don't need the dynamic range, > but you *do* need the volume, so be it: you would prefer the off-chance > of some clipping. It *has* to be a better solution to using the rxgain > setting if you don't need to: rxgain is going to affect echo for the > worse. Also notice that the volume of these files is sufficient when they > are played back over the telephone: it's only when you play them back via > a sound card that you have the volume problem. So, you can't just > willy-nilly amplify everything. I'd personally agree with every word above. There is a work around coming that will help with the VM gain issue while other work is in progress to identify the root cause. Might see the work around today if you monitor the cvs changes. ;) Rich ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] sip - aastra 9133i
Hello. Just rx'd the sip - aastra 9133i. Haven't done sip before. My initial attempt at setup has failed. "No Service" Anyone want to contact me off-list or on irc ? Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] Speaking of Polycom phones...updated ROM: ouch!
At 16:16 9/6/2005 -0700, Jesse Keating wrote: On Tue, 2005-09-06 at 17:41 -0500, Doug wrote: > After I did this, it appears that the Web interface > for the phone won't change the settings, nor will > it actually reboot the phone now. What do I have > to set on the phone itself, so I can update info > in the Web interface, and then restart the phone? > What you need to do is 'clear local config' before you start making changes. Menu -> Settings -> Advanced ( -> password ) -> Admin Settings -> Reset to Default -> Reset Local Config Once you've done that and rebooted, you should be able to make your changes through web or on the phone itself. Hey Jesse, Thanks for the advice. It worked like charm. Now I can set fields in the Web interface, and reboot. However, now I can't seem to the phone to register. Have you seen this before? I again followed instructions here: http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501 Would you be willing to save and zip up your config pages for the phone and Asterisk for me to compare with mine? http://192.168.2.5/netConf.htm http://192.168.2.5/appConf.htm http://192.168.2.5/reg.htm http://xxx.xxx.xx.xx/admin/config.php?display=3&extdisplay=10100 In the boot log, it shows something like: "x....cfg could not be downloaded. Getting next file..." That file does not exist on the TFTP server. Is this a problem? Also, in the boot log it shows DNS referring to 192.168.2.1, even though it's setup properly at the phone and its corresponding Web interface. Any other ideas? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM card and voicemail volume
[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM: > I was able to raise the volume from inaudible to acceptable by > increasing the RxGain in zapata.conf by 5db. I'd rather not go the > uncomressed wav route, as it will chew up storage in my email system. I know I'm way behind on reading this, but I thought I would follow up. According to this message: http://lists.digium.com/pipermail/asterisk-users/2004-November/072990.html the reason that uncompressed WAV files are louder is that the software that saves the WAV file is amplifying the volume of the files by shifting the data two bits to the left (or making it 4x louder). It is in no way "fixing" the underlying problem of the file being too quiet; it is just throwing away dynamic range in order to amplify the file. Now that may not be a bad solution: if you don't need the dynamic range, but you *do* need the volume, so be it: you would prefer the off-chance of some clipping. It *has* to be a better solution to using the rxgain setting if you don't need to: rxgain is going to affect echo for the worse. Also notice that the volume of these files is sufficient when they are played back over the telephone: it's only when you play them back via a sound card that you have the volume problem. So, you can't just willy-nilly amplify everything. Hope this helps. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.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] Occasional quiet voicemails
On Wednesday 07 September 2005 14:41, Rich Adamson wrote: > I don't believe 2023 has anything to do with the 411P; it was basically > an digium analog card issue (eg, TDM04b & x100p). > > Based on my tests and findings, the issue is the digium cards record > voicemail messages at a very low audio level (very different from > recording a voicemail from a sip phone). "If" the person leaving a VM > message called in via the digium card, and that user was located close > to "their" central office, the VM level is acceptable to poor. But, if > that same person is further from their central office (adding additional > transmission path loss), then that loss plus the digium analog card loss > makes the VM difficult if not impossible to hear. > > So, thinking that statement through very carefully, you might have some > users complain and other not, and the problem will not track against > anything that you have control over (eg, where the remote user is calling > from and the transmission loss they incur). > > If the digium analog cards passed audio through without any additional > loss, your user's probably would not be complaining. But that extra loss > is what I believe "is" the issue. > > Sounds like there might be a workaround coming for this. > > Rich zapata.conf usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=5.0 txgain=5.0 group=0 callgroup=1 pickupgroup=1 immediate=no improved it for me. YMMV. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk.org blocked - rejecting connections
On Wednesday 07 September 2005 13:47, Flobi wrote: > I'm not having any problems connecting to asterisk.org port 80. > They came up again. Finally. That check wasn't from where I am but another location once I couldn't get onto the site. Nothing more to see here...move on ;-0 Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.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] Occasional quiet voicemails
I don't believe 2023 has anything to do with the 411P; it was basically an digium analog card issue (eg, TDM04b & x100p). Based on my tests and findings, the issue is the digium cards record voicemail messages at a very low audio level (very different from recording a voicemail from a sip phone). "If" the person leaving a VM message called in via the digium card, and that user was located close to "their" central office, the VM level is acceptable to poor. But, if that same person is further from their central office (adding additional transmission path loss), then that loss plus the digium analog card loss makes the VM difficult if not impossible to hear. So, thinking that statement through very carefully, you might have some users complain and other not, and the problem will not track against anything that you have control over (eg, where the remote user is calling from and the transmission loss they incur). If the digium analog cards passed audio through without any additional loss, your user's probably would not be complaining. But that extra loss is what I believe "is" the issue. Sounds like there might be a workaround coming for this. Rich > Indeed I do - but I read bug 2023 before posting and thought it was to > do with the system-wide problem, not with occasional occurrences. I'll > go back and read it again. Has the problem been solved with the 411P? > > Regards, > -- > Anthony Rodgers > Business Systems Analyst > District of North Vancouver > Web: http://www.dnv.org > RSS Feed: http://www.dnv.org/rss.asp > > On Sep 6, 2005, at 7:32 PM, Rich Adamson wrote: > > > > Having solved a system-wide problem on 1.0.7 with low volume on > > > voicemail messages by using format=wav, some users are still > > > complaining that the occasional voicemail message (no apparent > > pattern > > > in terms of call origination) is still so quiet as to be barely > > > audible. Normal conversations and the majority of voicemail messages > > > are fine. > > > > > > Has anyone else experienced something similar? > > > > Yup. Bet you have an x100p or TDM card. See bug #2023 from a long > > time ago. > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.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 > > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk.org blocked - rejecting connections
I'm not having any problems connecting to asterisk.org port 80. On 9/7/05, Martin <[EMAIL PROTECTED]> wrote: > > Address lookup > canonical name asterisk.org. > aliases > addresses 216.27.40.102 > > Service scan > FTP - 21Error: TimedOut > SMTP - 25 Error: ConnectionRefused > HTTP - 80 Error: ConnectionRefused > POP3 - 110 Error: TimedOut > NNTP - 119 Error: TimedOut > > digium.com is ok though > > Address lookup > canonical name digium.com. > aliases > addresses 216.207.245.1 > > > Service scan > FTP - 21Error: TimedOut > SMTP - 25 Error: TimedOut > HTTP - 80 HTTP/1.1 302 Found > Date: Wed, 07 Sep 2005 18:38:11 GMT > Server: Apache > X-Powered-By: PHP/4.3.10 > Location: http://www.digium.com/ > Connection: close > Content-Type: text/html; charset=ISO-8859-1 > POP3 - 110 Error: TimedOut > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] asterisk.org blocked - rejecting connections
Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21Error: TimedOut SMTP - 25 Error: TimedOut HTTP - 80 HTTP/1.1 302 Found Date: Wed, 07 Sep 2005 18:38:11 GMT Server: Apache X-Powered-By: PHP/4.3.10 Location: http://www.digium.com/ Connection: close Content-Type: text/html; charset=ISO-8859-1 POP3 - 110 Error: TimedOut ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk crashed?
Which version of * are you using? I had a problem with 1.0.7 crashing unexplainably at one point, but 1.0.9 was out then and I upgraded and it stopped. On 9/7/05, Arik Funke <[EMAIL PROTECTED]> wrote: > Hi, > > I am running Asterisk in production mode but unfortunately every few > days or so, it crashes, presumably... > > Presumably because, when the phones stop working and I look for the > cause, asterisk is no longer running. Asterisk logs and > /var/log/messages contain no hints at all. > > How can I get mode info on such unpredicable crashes? > > Thanks in advance, > Arik > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
It's not that, it's just that the wiki wasn't very clear on the fact that all the tables for a static load had to be the same. I had thought that I was supposed to use the table on this page: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Extensions even when doing realtime static, which isn't the case, I had to use the table on http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static . Also, I wasn't trying to say the wiki was incorrect, just a little unclear. I didn't change any info, just added some clarification for those who might miss that part, like I did. On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > The wiki doc's are correct. You are trying to combine two different methods > of pulling RealTime extensions and that is why it isn't working as you are > expecting. > > Pick 1 method and all will be revealed. Both are very simple to do. > > -Matthew > > > From: Flobi <[EMAIL PROTECTED]> > > Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial > > Discussion > > Date: Wed, 7 Sep 2005 13:00:26 -0400 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [Asterisk-Users] Extensions - Realtime > > > > Nevermind, I figured out that the table is used way differently when > > doing static. Here's my fixed table. I'll try to explain this in the > > voip-info doc. > > > > id cat_metric var_metric commented filename category var_name > > var_val > > 1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing) > > > > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > >> Okay, after noticing an error on this mysql statement after i switched to > >> odbc: > >> SELECT * FROM > >> pbx_realtime_extensions > >> WHERE filename='extensions.conf' and commented=0 > >> ORDER BY filename,cat_metric desc,var_metric > >> asc,category,var_name,var_val,id > >> > >> I added those fields and reloaded...* immediately crashed. I > >> restarted. Now, I'm getting this: > >> *CLI> show dialplan > >> [ Context 'NoOp' created by 'pbx_config' ] > >> > >> [ Context 'parkedcalls' created by 'res_features' ] > >> '700' => 1. Park() > >> [res_features] > >> > >> -= 1 extensions (1 priorities) in 2 contexts. =- > >> > >> > >> out of this table: > >> id name context exten priority app appdata filename > >> commented cat_metric var_metric category var_name var_val > >> 1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL > >> NULL NULL NULL > >> > >> > >> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > >>> Okay, this doesn't seem to be working. I've gone and deleted my ael > >>> file also. I do know my MySQL is set up cause I have my sip, iax and > >>> voicemail going through it too. > >>> > >>> here's the line in extconfig.conf: > >>> [settings] > >>> extensions.conf => mysql,asterisk,pbx_realtime_extensions > >>> > >>> > >>> in pbx_realtime_extensions, my db table: > >>> id name context exten priority app appdata > >>> 1 default default _. 1 NoOp Testing > >>> > >>> > >>> CLI> show dialplan > >>> [ Context 'parkedcalls' created by 'res_features' ] > >>> '700' => 1. Park() > >>> [res_features] > >>> > >>> -= 1 extensions (1 priorities) in 1 contexts. =- > >>> > >>> And when I try to call, I get: > >>> Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected > >>> connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not > >>> exist > >>> > >>> Also, this message keeps popping up even when calls aren't going through: > >>> Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot > >>> find extension context 'default' > >>> > >>> > >>> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > > It states that the conf file overrides the static db info, but what about > the ael file? Does that override also? > > BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! > > Flobi > > > On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > >> CVS HEAD/Asterisk 1.2: Is there a way to have the entire > >> extensions.conffile coming from the realtime? > > > >Yes. Go read the wiki on RealTime Static. > > > > -Matthew > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing their > best to serve you and your indignant, malcontent attitude. > -- > >>> > >>> > >>> > >>>
RE: [Asterisk-Users] How to connect many analog lines to Asterisk?
Ohmy bad...I picked up the thread later :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josip Gracin Sent: Wednesday, September 07, 2005 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to connect many analog lines to Asterisk? Jonathan k. Creasy wrote: > You asked how to connect lines, so he answered that question. The > answer is basically the same just change the FXO in the channel bank > to FXS. Well, actually, I said: "If I have more than a hundred analog telephones (analog lines) that need..." But, that doesn't help my case, does it? :-) Anyway, thanks everybody for the info! ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
Jonathan k. Creasy wrote: You asked how to connect lines, so he answered that question. The answer is basically the same just change the FXO in the channel bank to FXS. Well, actually, I said: "If I have more than a hundred analog telephones (analog lines) that need..." But, that doesn't help my case, does it? :-) Anyway, thanks everybody for the info! ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
The wiki doc's are correct. You are trying to combine two different methods of pulling RealTime extensions and that is why it isn't working as you are expecting. Pick 1 method and all will be revealed. Both are very simple to do. -Matthew > From: Flobi <[EMAIL PROTECTED]> > Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial > Discussion > Date: Wed, 7 Sep 2005 13:00:26 -0400 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Extensions - Realtime > > Nevermind, I figured out that the table is used way differently when > doing static. Here's my fixed table. I'll try to explain this in the > voip-info doc. > > id cat_metric var_metric commented filename category var_name var_val > 1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing) > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: >> Okay, after noticing an error on this mysql statement after i switched to >> odbc: >> SELECT * FROM >> pbx_realtime_extensions >> WHERE filename='extensions.conf' and commented=0 >> ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id >> >> I added those fields and reloaded...* immediately crashed. I >> restarted. Now, I'm getting this: >> *CLI> show dialplan >> [ Context 'NoOp' created by 'pbx_config' ] >> >> [ Context 'parkedcalls' created by 'res_features' ] >> '700' => 1. Park() >> [res_features] >> >> -= 1 extensions (1 priorities) in 2 contexts. =- >> >> >> out of this table: >> id name context exten priority app appdata filename >> commented cat_metric var_metric category var_name var_val >> 1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL >> NULL NULL NULL >> >> >> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: >>> Okay, this doesn't seem to be working. I've gone and deleted my ael >>> file also. I do know my MySQL is set up cause I have my sip, iax and >>> voicemail going through it too. >>> >>> here's the line in extconfig.conf: >>> [settings] >>> extensions.conf => mysql,asterisk,pbx_realtime_extensions >>> >>> >>> in pbx_realtime_extensions, my db table: >>> id name context exten priority app appdata >>> 1 default default _. 1 NoOp Testing >>> >>> >>> CLI> show dialplan >>> [ Context 'parkedcalls' created by 'res_features' ] >>> '700' => 1. Park() >>> [res_features] >>> >>> -= 1 extensions (1 priorities) in 1 contexts. =- >>> >>> And when I try to call, I get: >>> Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected >>> connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not >>> exist >>> >>> Also, this message keeps popping up even when calls aren't going through: >>> Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot >>> find extension context 'default' >>> >>> >>> On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: It states that the conf file overrides the static db info, but what about the ael file? Does that override also? BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! Flobi On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: >> CVS HEAD/Asterisk 1.2: Is there a way to have the entire >> extensions.conffile coming from the realtime? > >Yes. Go read the wiki on RealTime Static. > > -Matthew > > > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- >>> >>> >>> >>> -- >>> Automated Signature: This message is from Flobi of Flobi.com. >>> Visit my website if you like: http://www.flobi.com/ >>> >>> Please remember to tip your waitress and bartender. They are doing >>> their best to serve you and your indignant, malcontent attitude. >>> -- >>> >> >> >> -- >> Automated Signature: This message is from Flobi of Flobi.com. >> Visit my website if you like: http://www.flobi.com/ >> >> Please remember to tip your waitress and bartender. They are doing >> their best to serve you and your indignant, malcontent attitude. >> -- >> > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing > their best to serve you and your indignant, malcontent attitude. > -- > ___
Re: [Asterisk-Users] Extensions - Realtime
I don't see your swich statement anywhere. You must define a context [default] then add in the correct switch=> statement. -Matthew > From: Flobi <[EMAIL PROTECTED]> > Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial > Discussion > Date: Wed, 7 Sep 2005 12:18:26 -0400 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Extensions - Realtime > > Okay, this doesn't seem to be working. I've gone and deleted my ael > file also. I do know my MySQL is set up cause I have my sip, iax and > voicemail going through it too. > > here's the line in extconfig.conf: > [settings] > extensions.conf => mysql,asterisk,pbx_realtime_extensions > > > in pbx_realtime_extensions, my db table: > id name context exten priority app appdata > 1 default default _. 1 NoOp Testing > > > CLI> show dialplan > [ Context 'parkedcalls' created by 'res_features' ] > '700' => 1. Park() > [res_features] > > -= 1 extensions (1 priorities) in 1 contexts. =- > > And when I try to call, I get: > Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected > connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not > exist > > Also, this message keeps popping up even when calls aren't going through: > Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot > find extension context 'default' > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: >> >> It states that the conf file overrides the static db info, but what about the >> ael file? Does that override also? >> >> BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! >> >> Flobi >> >> >> On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? >>> >>>Yes. Go read the wiki on RealTime Static. >>> >>> -Matthew >>> >>> >>> ___ >>> --Bandwidth and Colocation sponsored by Easynews.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 >>> >> >> >> >> >> -- >> Automated Signature: This message is from Flobi of Flobi.com. >> Visit my website if you like: http://www.flobi.com/ >> >> Please remember to tip your waitress and bartender. They are doing their >> best to serve you and your indignant, malcontent attitude. >> -- > > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing > their best to serve you and your indignant, malcontent attitude. > -- > ___ > --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
You asked how to connect lines, so he answered that question. The answer is basically the same just change the FXO in the channel bank to FXS. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josip Gracin Sent: Wednesday, September 07, 2005 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to connect many analog lines to Asterisk? Darren Wright wrote: > Wow, first of all, if you have a hundred analog lines, you are doing > yourself a disservice.a 4 T1's would be much much cheaper, and > much easier to manage. Let me clear this up a little bit. There are hundreds of telephone devices inside the building, all connected to a PBX, and there is an E1/T1 connection to the PSTN (being statistically multiplexed, obviously). What I'd like to do is to replace the PBX with Asterisk. I don't see how I can make the situation better by using 4 T1's? > Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and > fill them with FXO cards, and then buy 1 TE406 Quad T1 card for your > asterisk box. Thanks, I think that's what I need. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
On Wed, September 7, 2005 18:11, Josip Gracin said: > Darren Wright wrote: >> Wow, first of all, if you have a hundred analog lines, you are doing >> yourself a disservice.a 4 T1's would be much much cheaper, and much >> easier to manage. > > Let me clear this up a little bit. There are hundreds of telephone > devices inside the building, all connected to a PBX, and there is an > E1/T1 connection to the PSTN (being statistically multiplexed, > obviously). What I'd like to do is to replace the PBX with Asterisk. > > I don't see how I can make the situation better by using 4 T1's? > You said you had 100 analog lines... What you meant is you have 100 analog phones... Big difference... (OTOH, only a single letter: FXO -> FXS) ;-) But seriously, there really is a big difference whether you are trying to connect 100 analog lines (i.o.w. 100 incoming POTS lines from the PSTN) or 100 analog phones... If you had 100 incoming POTS lines, 4 PRI spans would be way cheaper and way easier, hence the advice! >> Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill >> them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk >> box. > > Thanks, I think that's what I need. > If you want 100 analog phones, make sure you get FXS cards in stead of FXO cards... FXO cards are for incoming lines, FXS for phones... (FXO stands for Foreign Exchange Office, ie PABX or PSTN, FXS for Foreign Exchange Subscriber, ie telephones) HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
Nevermind, I figured out that the table is used way differently when doing static. Here's my fixed table. I'll try to explain this in the voip-info doc. id cat_metric var_metric commented filename category var_name var_val 1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing) On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > Okay, after noticing an error on this mysql statement after i switched to > odbc: > SELECT * FROM > pbx_realtime_extensions > WHERE filename='extensions.conf' and commented=0 > ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id > > I added those fields and reloaded...* immediately crashed. I > restarted. Now, I'm getting this: > *CLI> show dialplan > [ Context 'NoOp' created by 'pbx_config' ] > > [ Context 'parkedcalls' created by 'res_features' ] > '700' => 1. Park() > [res_features] > > -= 1 extensions (1 priorities) in 2 contexts. =- > > > out of this table: > id name context exten priority app appdata filename > commented cat_metric var_metric category var_name var_val > 1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL > NULL NULL NULL > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > > Okay, this doesn't seem to be working. I've gone and deleted my ael > > file also. I do know my MySQL is set up cause I have my sip, iax and > > voicemail going through it too. > > > > here's the line in extconfig.conf: > > [settings] > > extensions.conf => mysql,asterisk,pbx_realtime_extensions > > > > > > in pbx_realtime_extensions, my db table: > > id name context exten priority app appdata > > 1 default default _. 1 NoOp Testing > > > > > > CLI> show dialplan > > [ Context 'parkedcalls' created by 'res_features' ] > > '700' => 1. Park() > > [res_features] > > > > -= 1 extensions (1 priorities) in 1 contexts. =- > > > > And when I try to call, I get: > > Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected > > connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not > > exist > > > > Also, this message keeps popping up even when calls aren't going through: > > Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot > > find extension context 'default' > > > > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > > > > > > It states that the conf file overrides the static db info, but what about > > > the ael file? Does that override also? > > > > > > BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! > > > > > > Flobi > > > > > > > > > On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > > > > > CVS HEAD/Asterisk 1.2: Is there a way to have the entire > > > > > extensions.conffile coming from the realtime? > > > > > > > >Yes. Go read the wiki on RealTime Static. > > > > > > > > -Matthew > > > > > > > > > > > > ___ > > > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > > > > > > > > > > > > > > > -- > > > Automated Signature: This message is from Flobi of Flobi.com. > > > Visit my website if you like: http://www.flobi.com/ > > > > > > Please remember to tip your waitress and bartender. They are doing their > > > best to serve you and your indignant, malcontent attitude. > > > -- > > > > > > > > -- > > Automated Signature: This message is from Flobi of Flobi.com. > > Visit my website if you like: http://www.flobi.com/ > > > > Please remember to tip your waitress and bartender. They are doing > > their best to serve you and your indignant, malcontent attitude. > > -- > > > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing > their best to serve you and your indignant, malcontent attitude. > -- > -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
Okay, after noticing an error on this mysql statement after i switched to odbc: SELECT * FROM pbx_realtime_extensions WHERE filename='extensions.conf' and commented=0 ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id I added those fields and reloaded...* immediately crashed. I restarted. Now, I'm getting this: *CLI> show dialplan [ Context 'NoOp' created by 'pbx_config' ] [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -= 1 extensions (1 priorities) in 2 contexts. =- out of this table: id name context exten priority app appdata filename commented cat_metric var_metric category var_name var_val 1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL NULL NULL NULL On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > Okay, this doesn't seem to be working. I've gone and deleted my ael > file also. I do know my MySQL is set up cause I have my sip, iax and > voicemail going through it too. > > here's the line in extconfig.conf: > [settings] > extensions.conf => mysql,asterisk,pbx_realtime_extensions > > > in pbx_realtime_extensions, my db table: > id name context exten priority app appdata > 1 default default _. 1 NoOp Testing > > > CLI> show dialplan > [ Context 'parkedcalls' created by 'res_features' ] > '700' => 1. Park() > [res_features] > > -= 1 extensions (1 priorities) in 1 contexts. =- > > And when I try to call, I get: > Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected > connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not > exist > > Also, this message keeps popping up even when calls aren't going through: > Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot > find extension context 'default' > > > On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > > > > It states that the conf file overrides the static db info, but what about > > the ael file? Does that override also? > > > > BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! > > > > Flobi > > > > > > On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > > > > CVS HEAD/Asterisk 1.2: Is there a way to have the entire > > > > extensions.conffile coming from the realtime? > > > > > >Yes. Go read the wiki on RealTime Static. > > > > > > -Matthew > > > > > > > > > ___ > > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > > > > > > > > > -- > > Automated Signature: This message is from Flobi of Flobi.com. > > Visit my website if you like: http://www.flobi.com/ > > > > Please remember to tip your waitress and bartender. They are doing their > > best to serve you and your indignant, malcontent attitude. > > -- > > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing > their best to serve you and your indignant, malcontent attitude. > -- > -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] ztcfg Kills My Dial Tone
I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS). These connect to a Digium TE210P card. I'm running kernel 2.6.10 and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today. The results are the same for all versions: Right after I reboot, and modprobe wct4xxp, my analog phone connected to port 13 of the first channel bank (first FXS port) gets a dial tone. Asterisk is not running yet, and I have NOT run ztcfg. As soon as I run ztcfg, the port goes dead. No dial tone, but I can hear things I say into the phone's microphone come out of the speaker, se there's voltage. Starting Asterisk makes no difference. The port is dead until I unload and reload the wct4xxp module. HOWEVER, when the port is dead, I can ring it from a SIP extension, and the analog phone rings. If I pick up the phone, Asterisk has no clue, and keeps ringing forever. My guess is this is a problem with the TE210P card or drivers. Any suggestions? *** Here's my /etc/zaptel.conf (all non comment lines): *** span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs fxsks=1-12 fxoks=13-48 defaultzone = us loadzone = us *** Here's the output of ztcfg -vvv: *** Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) Channel 17: FXO Kewlstart (Default) (Slaves: 17) Channel 18: FXO Kewlstart (Default) (Slaves: 18) Channel 19: FXO Kewlstart (Default) (Slaves: 19) Channel 20: FXO Kewlstart (Default) (Slaves: 20) Channel 21: FXO Kewlstart (Default) (Slaves: 21) Channel 22: FXO Kewlstart (Default) (Slaves: 22) Channel 23: FXO Kewlstart (Default) (Slaves: 23) Channel 24: FXO Kewlstart (Default) (Slaves: 24) Channel 25: FXO Kewlstart (Default) (Slaves: 25) Channel 26: FXO Kewlstart (Default) (Slaves: 26) Channel 27: FXO Kewlstart (Default) (Slaves: 27) Channel 28: FXO Kewlstart (Default) (Slaves: 28) Channel 29: FXO Kewlstart (Default) (Slaves: 29) Channel 30: FXO Kewlstart (Default) (Slaves: 30) Channel 31: FXO Kewlstart (Default) (Slaves: 31) Channel 32: FXO Kewlstart (Default) (Slaves: 32) Channel 33: FXO Kewlstart (Default) (Slaves: 33) Channel 34: FXO Kewlstart (Default) (Slaves: 34) Channel 35: FXO Kewlstart (Default) (Slaves: 35) Channel 36: FXO Kewlstart (Default) (Slaves: 36) Channel 37: FXO Kewlstart (Default) (Slaves: 37) Channel 38: FXO Kewlstart (Default) (Slaves: 38) Channel 39: FXO Kewlstart (Default) (Slaves: 39) Channel 40: FXO Kewlstart (Default) (Slaves: 40) Channel 41: FXO Kewlstart (Default) (Slaves: 41) Channel 42: FXO Kewlstart (Default) (Slaves: 42) Channel 43: FXO Kewlstart (Default) (Slaves: 43) Channel 44: FXO Kewlstart (Default) (Slaves: 44) Channel 45: FXO Kewlstart (Default) (Slaves: 45) Channel 46: FXO Kewlstart (Default) (Slaves: 46) Channel 47: FXO Kewlstart (Default) (Slaves: 47) Channel 48: FXO Kewlstart (Default) (Slaves: 48) 48 channels configured. *** Here are the kernel messages emitted when I run ztcfg -vvv: *** Sep 7 11:46:22 localhost kernel: About to enter spanconfig! Sep 7 11:46:22 localhost kernel: About to enter startup! Sep 7 11:46:22 localhost kernel: wct2xxp: Setting yellow alarm on span 1 Sep 7 11:46:22 localhost kernel: Zaptel: Master changed to TE2/0/2 Sep 7 11:46:22 localhost kernel: TE2XXP: Span 1 configured for ESF/B8ZS Sep 7 11:46:22 localhost kernel: Putting 0 in register 2f on span 1 Sep 7 11:46:22 localhost kernel: Putting 0 in register 30 on span 1 Sep 7 11:46:22 localhost kernel: Putting 0 in register 31 on span 1 Sep 7 11:46:22 localhost kernel: SPAN 1: Primary Sync Source Sep 7 11:46:22 localhost kernel: Completed startup! Sep 7 11:46:22 localhost kernel: About to enter spanconfig! Sep 7 11:46:22 localhost kernel: About to enter startup! Sep 7 11:46:22 localhost kernel: wct2xxp: Setting yellow alarm on span 2 Sep 7 11:46:22 localhost kernel: TE2XXP: Span 2 configured for ESF/B8ZS Sep 7 11:46:22 localhost kernel: Putting 0 in register 2f on span 2 Sep 7 11:46:22 localhost kernel: Putting 0 in register 30 on span 2 Sep 7 11:46:22 localhost kernel: Putting 0 in register 31 on span 2 Sep 7 11:46:22 localhost kernel: Completed startup! Sep 7 11:46:22 localhost k
[Asterisk-Users] Second Line does not Connect - HELP - misdn,sip
About my System:
2 * HFC Cards with misdn. 1 NT mode, 1 TE mode
1 * Sip-Provider (1und1)
On NT-Port à Ritto
(Elmeg) PBX
On TE-Port à NTBA
About my Problem:
When a SIP-Call from a phone connected to the Ritto PBX is
in progress and someone calls on the ISDN-Line, the greeting works, and the phones
connected to the Ritto PBX are ringing. When I pick up a phone there is no connection
and the caller hears MOH all the time. This only happens when a second call is
in progress.
When no other call is in progress, everything works fine.
About my Configfiles:
extensions.conf
[incoming]
exten => ,1,Goto(anruferannahme,s,1)
exten => ,1,Goto(anruferannahme,s,1)
exten => 922xxx,1,Answer()
exten => 922xxx,2,Dial(misdn/2/922975) ; FAX
exten => 923xxx,1,Answer()
exten => 923xxx,2,Playback(thomas)
exten => 923xxx,3,Dial(misdn/2/9230250,,m) ; Thomas
Durchwahl
exten => 923xxx,1,Answer()
exten => 923xxx,2,Dial(misdn/2/9230251) ; Thomas FAX
[outgoing]
; Anwahl über normale ISDN-Leitung:
exten => _999.,1,Answer()
exten => _999.,2,Dial(misdn/1/${EXTEN:3},,m)
exten => _999.,3,Playback(dialfailed)
; Faxe über normalen ISDN-Anschluss verschicken:
exten => _X./922975,1,WaitforDigits(2000) ; mit
Vorwahl
exten => _X./922975,2,Answer()
exten => _X./922975,3,Dial(misdn/1/${EXTEN}) ; wenn IP
nich erfolgreich
; Telefongespräche bei denen die Vorwahl angegeben ist:
exten => _0X.,1,WaitforDigits(4000)
exten => _0X.,2,Answer()
exten => _0X.,3,Dial(SIP/[EMAIL PROTECTED])
exten => _0X.,4,Playback(nosip)
exten => _0X.,5,Dial(misdn/1/${EXTEN}) ; wenn IP nicht
erfolgreich
exten => _0X.,6,Playback(dialfailed)
exten => _0X.,104,Playback(besetzt)
; Telefongespräche bei denen die Vorwahl nicht
angegeben ist:
exten => _X.,1,WaitforDigits(4000)
exten => _X.,2,Answer()
exten => _X.,3,Dial(SIP/[EMAIL PROTECTED])
exten => _X.,4,Playback(nosip)
exten => _X.,5,Dial(misdn/1/${EXTEN})
exten => _X.,6,Playback(dialfailed)
exten => _X.,104,Playback(besetzt)
[aufnahme]
exten => s,1,Background(beep)
exten => 1,1,Record(/var/lib/asterisk/sounds/greeting:gsm)
exten => 2,1,Record(/var/lib/asterisk/sounds/besetzt:gsm)
exten =>
3,1,Record(/var/lib/asterisk/sounds/aufnahme:gsm)
[anruferannahme]
exten => s,1,Answer()
exten => s,2,Background(greeting)
exten => s,3,Dial(misdn/2/,15,m)
;exten => s,4,WaitMusicOnHold(2)
;exten => s,5,Dial(misdn/2/9230255,15,m)
;exten => s,6,WaitMusicOnHold(2)
;exten => s,7,Dial(misdn/2/,100,m)
;exten => s,8,Playback(nichterr)
exten => s,4,Hangup()
exten => 7,1,Goto(aufnahme,s,1)
misdn.conf
[general]
context=vs
language=de
immediate=yes
debug=2
allow=alaw
musiconhold=default
[TEport]
context=incoming
ports=1
msns=*
[NTport]
context=outgoing
ports=2
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
externip = myip
localnet = 192.168.0.0/255.255.0.0
context = default
srvlookup = yes
disallow = all
allow = ulaw
nat = yes
register => 492774:[EMAIL PROTECTED]/492774
[sip.1und1.de]
type=friend
username=492774
fromuser=492774
secret=mysecret
host=sip.1und1.de
context=incoming
fromdomain=1und1.de
qualify=no
insecure=very
canreinvite=no
nat=yes
allow=g726
dtmfmode=info
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[Asterisk-Users] Asterisk crashed?
Hi, I am running Asterisk in production mode but unfortunately every few days or so, it crashes, presumably... Presumably because, when the phones stop working and I look for the cause, asterisk is no longer running. Asterisk logs and /var/log/messages contain no hints at all. How can I get mode info on such unpredicable crashes? Thanks in advance, Arik ___ --Bandwidth and Colocation sponsored by Easynews.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] Working example of ALERT_INFO with Cisco ATAs?
Olle E. Johansson wrote: Try setting _ALERT_INFO Worked perfectly, thanks. B. ___ --Bandwidth and Colocation sponsored by Easynews.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] Lock Extension
Hi Robert, Do you have the sample script for locking the extension? Thanks, Stephen Robert Goodyear wrote: On Aug 18, 2005, at 3:07 AM, Stephen wrote: Hi All, How can I lock the extension in Asterisk? For example , my extension is 1000 and I am away for business trip. I want to lock my extension during my absence. Can it be done in Asterisk? regards, Stephen You could write a little script to mangle/unmangle your SIP context and then SIP RELOAD. You could assign it to a context called 'disabled' whose only valid extension matching therein is to that same macro to authenticate and change your context back. ___ 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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
Okay, this doesn't seem to be working. I've gone and deleted my ael file also. I do know my MySQL is set up cause I have my sip, iax and voicemail going through it too. here's the line in extconfig.conf: [settings] extensions.conf => mysql,asterisk,pbx_realtime_extensions in pbx_realtime_extensions, my db table: id name context exten priority app appdata 1 default default _. 1 NoOp Testing CLI> show dialplan [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -= 1 extensions (1 priorities) in 1 contexts. =- And when I try to call, I get: Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not exist Also, this message keeps popping up even when calls aren't going through: Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot find extension context 'default' On 9/7/05, Flobi <[EMAIL PROTECTED]> wrote: > > It states that the conf file overrides the static db info, but what about the > ael file? Does that override also? > > BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! > > Flobi > > > On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > > > CVS HEAD/Asterisk 1.2: Is there a way to have the entire > > > extensions.conffile coming from the realtime? > > > >Yes. Go read the wiki on RealTime Static. > > > > -Matthew > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.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 > > > > > > > -- > Automated Signature: This message is from Flobi of Flobi.com. > Visit my website if you like: http://www.flobi.com/ > > Please remember to tip your waitress and bartender. They are doing their > best to serve you and your indignant, malcontent attitude. > -- -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Re: Polycom 300 with latest 1.5.3 firmware not registering
Hi Jorge - I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I compared to a Grandstream registration, and it tries to REGISTER to the asterisk receiving a 200 Message response and effectively registering. Finding in the packet capture no other great difference, I believe that SUBSCRIBE requires a different authentification approach, maybe related to the voIpProt.SIP.requestValidation.digest.realm parameter in sip.cfg. I Tried the Polycom default, empty, default (voicemail context), from-internal (Extension context), the IP of the asterisk box, the name of the asterisk box, asterisk, etc, with no result. I tried different approaches documented in the wiki and related pages with no result. I can makke calls but I cannot receive them. To start off, can you post your Polycom phone.cfg file and your asterisk sip.conf files? Thanks, Noah ___ --Bandwidth and Colocation sponsored by Easynews.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] Occasional quiet voicemails
Indeed I do - but I read bug 2023 before posting and thought it was to do with the system-wide problem, not with occasional occurrences. I'll go back and read it again. Has the problem been solved with the 411P? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Sep 6, 2005, at 7:32 PM, Rich Adamson wrote: > Having solved a system-wide problem on 1.0.7 with low volume on > voicemail messages by using format=wav, some users are still > complaining that the occasional voicemail message (no apparent pattern > in terms of call origination) is still so quiet as to be barely > audible. Normal conversations and the majority of voicemail messages > are fine. > > Has anyone else experienced something similar? Yup. Bet you have an x100p or TDM card. See bug #2023 from a long time ago. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
Darren Wright wrote: Wow, first of all, if you have a hundred analog lines, you are doing yourself a disservice.a 4 T1's would be much much cheaper, and much easier to manage. Let me clear this up a little bit. There are hundreds of telephone devices inside the building, all connected to a PBX, and there is an E1/T1 connection to the PSTN (being statistically multiplexed, obviously). What I'd like to do is to replace the PBX with Asterisk. I don't see how I can make the situation better by using 4 T1's? Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk box. Thanks, I think that's what I need. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] ArtDio IPF-2000 unable to send audio to Cisco 7940 until placed on hold and resumed
The issue appears to be between the Cisco 7940 and the ArtDio IPF-2000, when a call is initiated between these phones the ArtDio phone receives the audio stream fine from the Cisco, but the Cisco can’t hear anything from the ArtDio, until the Cisco user places the call on hold and then picks the line back up. When the Cisco user places the call on hold, music streams fine from the server to the ArtDio and is played. Incoming/outgoing calls through the Asterisk server with the PSTN network are flawless with either the ArtDio or Cisco phone, and calls between ArtDio-only or Cisco-only are flawless. What gives?? I tried setting specific audio codecs in sip.conf for each phone with disallow=all and then allow=ulaw, I set the SIP.cnf file for the Cisco phone to be 711u as the preferred codec, and through the web interface configured the ArtDio to use 711u as it’s preferred codec, to no avail. I tcpdump’ed the data during the call and set the ArtDio phone to use RTP port and Control ports within the 17000 – 32000 range (SIP.cnf sets the Cisco phones to use ports dynamically from this range, based on tcpdump analysis) We’re using Asterisk 1.0.9 (Xorcom Rapid 1.1 latest release). I can post further configuration files if required. Any pointers would be much appreciated! -- Michael Coburn Network Solutions Manager MidWest Technical Associates c: 614-425-9203 p: 614-336-3640 x501 f: 614-336-3645 www.midwesttechnical.net ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
It states that the conf file overrides the static db info, but what about the ael file? Does that override also? BTW, "RealTime Static"...talk about oxymoron :-) Gotta love it! Flobi On 9/7/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > CVS HEAD/Asterisk 1.2: Is there a way to have the entire> extensions.conffile coming from the realtime? Yes. Go read the wiki on RealTime Static.-Matthew___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Automated Signature: This message is from Flobi of Flobi.com.Visit my website if you like: http://www.flobi.com/Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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] PRI in and out
Sorry my mistake. The span to provider is pri_cpe, and the span to the avaya is pri_net. On 9/7/05, Rod Bacon <[EMAIL PROTECTED]> wrote: > It DOES help, thanks. > > Except for this > > > > the only difference between the first set of channels (1-23) and the > > second set of channels (25-47) is: > > signalling=pri_net > > group=1 > > context = fromprovider > > channel => 1-23 > > signalling = pri_cpe > > group=2 > > context=fromavaya > > channel=25-47 > > I thought the signalling setting was from the perspective of the * server, not > the other side. For example, my PRIs to my provider are configured as pri_cpe, > as I am the CPE. > > Your example seems to suggest the other way around. > > ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
I don't know why Darren syas 3 Adits since each one can handle 48 FXO/FXS channels, so 2 make 96. Anyhow each Adit connects to 2 T1 ports on a TE405/6. With Adit 600 I don't see why TE406 is required since I believe the Adit 600 will take care of the echo, I might be wrong on this last one about the echo. In any case, if what you want is FXS cards, then I would strongly recommend to get Adit 600 with CMG cards, then you can get by with a $400 PC for asterisk, since you don't need any digium cards for that, because the CMG in the Adit will handle the main load of transcoding. The one limitation is, that if you need faxing, or FXO ports then don't go with the CMG. It might still be worthwhile to get a single span T1 card from Digium, use an Adit to cross connect the FXS cards/channels that are connected to fax machines to that T1, as well as any FXO cards that need to go to asterisk. On 9/7/05, Darren Wright <[EMAIL PROTECTED]> wrote: > Wow, first of all, if you have a hundred analog lines, you are doing > yourself a disservice.a 4 T1's would be much much cheaper, and much > easier to manage. > > Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill > them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk > box. > > -Darren > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Josip > Gracin > Sent: Wednesday, September 07, 2005 5:20 AM > To: [EMAIL PROTECTED]: [Asterisk-Users] How to > connect many analog lines to Asterisk? > > Hello! > > If I have more than a hundred analog telephones (analog lines) that need > > to be connected to Asterisk PBX, what kind of hardware do I need, and > where can I buy it? > > Thanks in advance! > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > > > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] PBX Replacement
I am getting ready to spec out a replacement for a Merlin Legend system with asterisk. There are a couple of things that holding me up that hopefully someone here can answer. 1. How well do modems work through a channel back to a PRI/T1 interface? 2. Is there a decent receptionist phone (I don't want to use FOP) to replace the system that our receptionists are already familiar with? I know that there is some discussion on the cisco 7914s and the SNom side car... but do they work? and how well... 3. Network suggestions... this is more open ended.. we are currently 100mb to the desktop, however the switches are garbage and I have heard that it is best to vlan the voice traffic away from the data traffic... thoughts? Regards, Sean Cook Network Engineer Kinex Networking ___ --Bandwidth and Colocation sponsored by Easynews.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] Speex codec - Out of buffer space
> I'm running Asterisk 1.0.7 and would like to add Speex support. I > downloaded Speex 1.0.5, installed and recompile Asterisk again. > > Now trying from X-Lite to connect using Speex but getting lot of weird > erros on Asterisk console: > > Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: > Out of buffer space > > I was trying to setup Speex on my second Asterisk server and wanted to > use this codec for IAX between these two boxes. But I'm getting unable > to negotiate codecs. Other codecs works like a charm. v1.0.7 is pretty old. Current cvs-head has speex built in. Would suggest upgrading asterisk code. ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM400 and Phone does not 'ring'
> 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it > does 'ring-ri... ri ring... ri...) > and the 3rd one does not ring at all when Asterisk says 'Ringing Zap/6'. > However, when I do an 'off-hook' on this phone, I get tone signal and > can dial and talk perfectly. > > I have phones compliant to the Belgium (Belgacom) Telco specs. > Are there differences in 'Ring Voltage' ? There is an issue here in France with our Siemens DECT phones that required a patch to change the ring _frequency_. It was given here ages ago, but now I can't find it. I believe it is still in "bugs". It requires a change to one line in wcfxs.c ___ --Bandwidth and Colocation sponsored by Easynews.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] Extensions - Realtime
> CVS HEAD/Asterisk 1.2: Is there a way to have the entire > extensions.conffile coming from the realtime? Yes. Go read the wiki on RealTime Static. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.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] PHP and ASterisk Manager
I fixed the problem using preg_replace but you are right, I completely
forgot We are using open source ! :) silly of me, I should have checked
that.
Thx for reopening my eyes Christoph
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Christoph Eicke
|Sent: Miércoles, 07 de Septiembre de 2005 02:12 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
|
|I looked into the source code of Asterisk to figure out how
|the printf() statements were spaced. That's the power of open
|source, you can look under the hood for these questions. It's
|easy to find, even for non-C-Gurus. Just do a "grep" for the
|string that you want inside of the Asterisk source directory
|and it will give you the file that the string you are looking
|for is in. Then simply open the file, search for the string
|and look at the
|printf() statement.
|
|Christoph
|
|On Tuesday 06 September 2005 21:16, Anton Krall wrote:
|> I was able to do and if and while loops to get the block of
|lines I want..
|> Now.. Another issue.
|>
|> I need to parse the line read to insert it into a table but seems
|> Asterisk inserts TABS or SPACES inconsistantly.. For example:
|>
|> Xxx(TAB)xxx(5 spaces)xxx
|> Next line
|> Xxx(TAB)xxx(3 spaces)xxx
|>
|> Im having a hard time figuring out how Asterisk Manager returns the
|> stuff
|> :)
|>
|> Well..s o far so good...
|>
|> |-Original Message-
|> |From: [EMAIL PROTECTED]
|> |[mailto:[EMAIL PROTECTED] On Behalf
|Of Matthew
|> |Boehm
|> |Sent: Martes, 06 de Septiembre de 2005 01:49 p.m.
|> |To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
|> |
|> |Anton Krall wrote:
|> |> Guys, is anybody using PHP sockets to connect to the Manager
|> |
|> |and send
|> |
|> |> command like "show voicemail users" for example or any other?
|> |>
|> |> My question is, how to parse the return info in a way that can be
|> |> shown back to the user via web (discard all the manager
|> |
|> |responses not needed)?
|> |
|> |Use preg_match() to match the lines you want the user to see on the
|> |website.
|> |
|> |$socket = fsockopen("localhost","5038", $errno, $errstr, 30);
|> |
|> |if(!$socket) {
|> | print "No socket";
|> |exit();
|> |}
|> |
|> |fputs($socket, "Action: Login\r\n");
|> |fputs($socket, "Events: Off\r\n");
|> |fputs($socket, "UserName: bleh\r\n"); fputs($socket, "Secret:
|> |bleh\r\n\r\n");
|> |
|> |fputs($socket, "Action: Command\r\n"); fputs($socket,
|"Command: show
|> |channels\r\n\r\n");
|> |
|> |fputs($socket, "Action: Logoff\r\n\r\n");
|> |
|> |while(!feof($socket)) {
|> | $buff = fgets($socket,1024);
|> | if(preg_match("/SIP\/.*/", $buff)) {
|> |print "I found a SIP call";
|> | }
|> |}
|> |
|> |___
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|> |
|> |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
|>
|> ___
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|>
|> 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
|___
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|
|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
|
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RE: [Asterisk-Users] Extensions - Realtime
I’ve got some modifications I’ve made to asterisk that create a “global switch”. It essentially just adds a check to the end of pbx_find_extension() that will try to look the extension up in the database if it’s not found in one of the includes or in any of the switches attached to the context itself. It’s rather hackish (it uses a global context not linked in with the regular context list), and so probably has some issues, but I can clean it up and post the patch somewhere if others are interested. It sounds like you would be. Cheers! Robert Bedell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Flobi Sent: Wednesday, September 07, 2005 9:57 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Extensions - Realtime CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conf file coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files. I'm using a PHP w/ MySQL AGI app to route my calls and all this would all be find except for blind transfers as I could use the channel variable itself to determine the source of the call. But, the blind transfer comes into the dialplan on the channel of the transferree with the context of the transferer. So, I need to have each phone in it's own context to determine what channel is actually requesting the transfer (for cdr, parking, routing, etc. reasons). As such, I have to create a new context each time I add a phone. I don't mind reloading so much and it looks like I'm not going to be able to avoid that anyways with the SIP RealTime cached (<--oxymoron) for MWI. The reason I don't want to edit files is that I'm sharing the dialplan between multiple boxes (the PHP app takes care of figuring out which box it is). I don't want to have to a. have to save the file on each box or b. map the files between boxes. !!! - Alternatively, if there is a way to determine in the dialplan who is the transferer without having each phone in it's own context, that would be fine. -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] CONNECT ACK timeout in libpri
Hi I am testing a voip gateway product with Asterisk. We are experiencing CONNECT ACK timer (T313) timing out on the Asterisk side when an incoming call is received on the T1-PRI interface. The call is immediately routed to voice mail. This doesn't happen if I connect another PRI test equipment to the voip gateway. The T313 timer is defined as 4000 mS however we see libpri complaining (i.e., timing out) immediately after sending CONNECT to the voip gateway. It seems it is not waiting for 4 seconds as before timing out. Any ponters would be appreciated. Regards GS ___ --Bandwidth and Colocation sponsored by Easynews.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] Which Linux distribution?
I use Centos 3.5 with great success. It is a RHEL3 binary compatible clone. Don't know if I would use development cutting edge software in the enterprise. --- John Daragon <[EMAIL PROTECTED]> wrote: > Tzafrir Cohen wrote: > > On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote: > > > >>YT Lim wrote: > >> > >>>We have tried Asterisk 1.0.9 on FC4 and have never > >>>been able to get CAPI (with Fritz card, fcpci) to work > >>>properly. Apart from that Asterisk works fine in > >>>switching internal calls. But it's useless if we can't > >>>make outgoing calls on our ISDN line. > >>> > >>>We are considering abandoning FC4 for Debian or SuSe. > >>>What is the general concensus on the best Linux to run > >>>Asterisk with CAPI? > >> > >>SUSE (as far as I know) is the only distro that really *expects* you to > >>be using ISDN2e as a matter of course. > > > > > > "Only Linux distro that" is generally something that is a bit hasty to > > say, given the fact that there are so many of them ;-) . > > You're absolutely right. > > > Mandrake is quite Europe-centric as well. I'm not sure about ISDN > > support. > > It's shipped with the packages; I looked at it when I first started > installing *, but couldn't get fcpci to work at the time. CAPI appears > to have been written on (or for) SUSE in the first place, and SUSE was > the first distro I came across that supported ISDN2e out of the box. > > > > > Debian has generally a large european installed base and a variety of > > ISDN-related packages as a result. > > > > Sorry, I won't make your life easier :-p > > > > You mean it's *supposed* to be easy ? > > jd > -- > > John Daragon [EMAIL PROTECTED] > argv[0] limited > Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK > v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 > > > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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] channels VHF/ HF radio in asterisk
The VHF or HF is determined by the radio equipment you have attached, not the software. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy Sent: Wednesday, September 07, 2005 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] channels VHF/ HF radio in asterisk Is then possible using app_rpt solution for both VHF and HF channels? Regards. Mark Phillips escribió: > 2 ways. > > 1) buy into the app_rpt system. They have a bespoke card for your PC > that can drive radio's etc. It's mainly aimed at repeater owners. > > 2) connect a phone patch between an ATA and your HF rig. This solution > is currently being used to provied phone services from a few Red Cross > shelters to the ARC HQ in Montgomery, AL. It works well. > > Mark, KC2ENI > > makevuy wrote: > >> Hy, >> I have a network with WIFI communication and VHF/ HF channels. I have >> integrated asterisk in the network using SIP, ZAP and IAX2 channels >> for WIFI communications, but I don't Know How I could integrate the >> VHF/ HF channels. >> >> I have heard speaking about app_rpt project, but I don't Know very >> much about this. >> >> Could I integrate VHF/ HF channels with this application? if the >> answer is yes, How? >> >> Regards. >> >> ___ >> --Bandwidth and Colocation sponsored by Easynews.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 >> > ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Re: ISDN PBX integration
Well, I just answer myself here: Since the ISDN PBX is just the same as ISDN phone as far as the asterisk is concerned, NT mode on the ISDN card should be used as well. The difference is that the phone uses p2mp (point to multi point) protocol, as the PBX uses p2p (point to point) protocol. Using the bri_net signalling instead of bri_net_ptmp solves the problem. Shahar On 9/7/05, Shahar Livne <[EMAIL PROTECTED]> wrote: > Hello list, > > I am trying to connect an old ISDN PBX to my asterisk system. > The setup includes an asterisk (1.0.9) running on the Soekris > hardware, with an ISDN card (Billion BIPAC PCI), and I run > zaphfc-bristuff-0.2.0-RC8k kernel module in NT mode (modes=1). > > When I connect an ISDN phone to the card (using cross ISDN cable + 40v > power supply), I manage to make calls from and to the asterisk. > > When I try to connect the ISDN PBX instead, still using the same cross > ISDN cable and the power supply, I get rings to the right extensions > of the ISDN PBX, but no call setup happens, and after 2 rings comes > silence. > > I have tried this with NT and TE modes of the zaphfc kernel module (I > thought it has to do with point-to-point or point-to-multipoint > issue). I have tried it both with cross ISDN cable and normal ISDN > cable (I just used Ethernet cable), all without any success (not even > the mentioned rings). > > When this ISDN PBX is connected to normal ISDN line, it just works. > > If there is anyone with some knowledge about ISDN equipment and its > integration with asterisk – I will be happy to get some advice – what > to check or how is it actually supposed to work. > > Thanks, > Shahar > > -- > Shahar Livne > LivneX - Open Source Development and Services > http://LivneX.com > -- Shahar Livne LivneX - Open Source Development and Services http://LivneX.com ___ --Bandwidth and Colocation sponsored by Easynews.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] channels VHF/ HF radio in asterisk
Is then possible using app_rpt solution for both VHF and HF channels? Regards. Mark Phillips escribió: 2 ways. 1) buy into the app_rpt system. They have a bespoke card for your PC that can drive radio's etc. It's mainly aimed at repeater owners. 2) connect a phone patch between an ATA and your HF rig. This solution is currently being used to provied phone services from a few Red Cross shelters to the ARC HQ in Montgomery, AL. It works well. Mark, KC2ENI makevuy wrote: Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is yes, How? Regards. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Extensions - Realtime
CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conf file coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files. I'm using a PHP w/ MySQL AGI app to route my calls and all this would all be find except for blind transfers as I could use the channel variable itself to determine the source of the call. But, the blind transfer comes into the dialplan on the channel of the transferree with the context of the transferer. So, I need to have each phone in it's own context to determine what channel is actually requesting the transfer (for cdr, parking, routing, etc. reasons). As such, I have to create a new context each time I add a phone. I don't mind reloading so much and it looks like I'm not going to be able to avoid that anyways with the SIP RealTime cached (<--oxymoron) for MWI. The reason I don't want to edit files is that I'm sharing the dialplan between multiple boxes (the PHP app takes care of figuring out which box it is). I don't want to have to a. have to save the file on each box or b. map the files between boxes. !!! - Alternatively, if there is a way to determine in the dialplan who is the transferer without having each phone in it's own context, that would be fine. -- Automated Signature: This message is from Flobi of Flobi.com.Visit my website if you like: http://www.flobi.com/Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.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] How to connect many analog lines to Asterisk?
Wow, first of all, if you have a hundred analog lines, you are doing yourself a disservice.a 4 T1's would be much much cheaper, and much easier to manage. Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk box. -Darren -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josip Gracin Sent: Wednesday, September 07, 2005 5:20 AM To: [EMAIL PROTECTED]: [Asterisk-Users] How to connect many analog lines to Asterisk? Hello! If I have more than a hundred analog telephones (analog lines) that need to be connected to Asterisk PBX, what kind of hardware do I need, and where can I buy it? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
RES: [Asterisk-Users] Billing - Disable accounts when balance gets0 value
[EMAIL PROTECTED] Simoni, Thank you for your copersation. If you need routes in Brazil I have very high quality ones ok... Atenciosamente Reduzimos ao mínimo a sua conta de Telefone Liguetel - ITN Info - 15 anos em Telecomunicações Diretoria Comercial - Newton Medina PABX11.3891.2434 Fax 11.38980112 msn [EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil Visite a Loja www.liguetel.com.br ou www.liguetel.com e conheça produtos e serviços para reduzir definitivamente a sua conta de telefone. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Simone Cittadini Enviada em: segunda-feira, 5 de setembro de 2005 05:54 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Billing - Disable accounts when balance gets0 value >This billing is also able to set accounts balance and for each call. Now I >need to disable accounts which balance gets a determined value. I was >thinking on changing account pass for that specif account which we need to >disable. And then in the sip.com reload info. > >Can you help me with new (new ways for doing so) or programing ideas too >once billing server has not the same public IP than Asterisk server. I ll >appreciate your comments ok. > > > I use ser+radius to do authentication, this way I can disable users or groups of users in a "standard" way, without using tricks like changing passwords. (when your customer pays he expect to have the same password as before, have you saved it ? where ? in a safe way ?) radius has a mysql backend, so also no need to reload config files. Asterisk and radius share the same db, with some not-too-complex agi before the actual Dial you can do stuff like setting the call timeout based on the remaining credit, blocking the call if the credit is too much in the red, and so on... ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/89 - Release Date: 2/9/2005 ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk overheating on VIA Epia M Seriesmotherboard
Angus - I have several mini-itx systems based on the Epia MII6000 (fanless) system. They all run great, and I have no problems. I also run 'mpg123' with several mp3s. I run it in an embedded configuration (in house). However, I do remember one board that I got where the heatsink on the CPU was loose which caused the thermal compound to be detached from the CPU. I removed the heatsink and put a silver compound in the place of the other compound, and we were okay again. My systems usually run around 45C->50C under load. Angus Comber wrote: But the systems are sold in this configuration. There is a fan option. I chose the fanless option. Angus - Original Message - From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, September 06, 2005 1:28 AM Subject: Re: [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard As you suspected, the problem is the fact that you don't have a fan. Since a machine that runs just a file server does not require much CPU power, the CPU doesn't get too hot. However Asterisk does use lots of CPU, therefore the CPU is hot, and yes the problem of stopping to work is because of the CPU being overheated, you are lucky that the computer booted after that, in most cases the overheating of a CPU means that the CPU expanded too much, when you shut it down it cools off, and shrinks, which could result in cracking the CPU. You should never run a CPU without it's fan if it's meant to run with a fan. Even if running it just as a file server. The fact that you are lucky doesn't mean that you don't need a fan. On 9/5/05, Angus Comber <[EMAIL PROTECTED]> wrote: Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file server and that does not run so hot and does not overheat? Why the difference? Just booting up both systems for 15 minutes you can tell the Asterisk box is quite a bit hotter. Also the Asterisk box overheated (well think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus __ ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Polycom 300 with latest 1.5.3 firmware not registering
Hello, I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I compared to a Grandstream registration, and it tries to REGISTER to the asterisk receiving a 200 Message response and effectively registering. Finding in the packet capture no other great difference, I believe that SUBSCRIBE requires a different authentification approach, maybe related to the voIpProt.SIP.requestValidation.digest.realm parameter in sip.cfg. I Tried the Polycom default, empty, default (voicemail context), from-internal (Extension context), the IP of the asterisk box, the name of the asterisk box, asterisk, etc, with no result. I tried different approaches documented in the wiki and related pages with no result. I can makke calls but I cannot receive them. I've seen mails stating that some installations have more than 100 phones working perfectly, can someone point me in the right direction to solve this ? Regards, Jorge A. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Speex codec - Out of buffer space
Hi, I'm running Asterisk 1.0.7 and would like to add Speex support. I downloaded Speex 1.0.5, installed and recompile Asterisk again. Now trying from X-Lite to connect using Speex but getting lot of weird erros on Asterisk console: Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: Out of buffer space I was trying to setup Speex on my second Asterisk server and wanted to use this codec for IAX between these two boxes. But I'm getting unable to negotiate codecs. Other codecs works like a charm. Any ideas? Thank you. -- David ___ --Bandwidth and Colocation sponsored by Easynews.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
