Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5
G.Day! Thanks for the response! i've tryed to do this, but in /var/spool/hylafax/log/xferfaxlog I read this: 09/06/11 09:04 CALL00108 ttyIAXfax +39.06.456789 0 0 0:00:09 0:00:09 Failure to receive silence (synchronization failure). 06654321 NONE::s what is it?! -- From: Larry Moore lmo...@starwon.com.au Sent: Monday, September 05, 2011 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5 On 5/09/2011 10:05 PM, Alessio wrote: someone can help me to solve this problem? thanks -- From: Alessio ales...@asistar.it Sent: Friday, September 02, 2011 5:10 PM To: Lee Howard fax...@howardsilvan.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5 1: from the phone i called the fax-server 2: from external fax i tried to send a fax to fax-server the results: _ G'Day Alessio, I replied to your original post suggesting you set up two IAX modems and get successful transmission working between them. I suspect you want to use T.38 with IAX modem, I don't believe the IAX2 channel supports T.38 hence I would suggest you remove the t38pt_udptl lines from your iax.conf files to avoid confusion. I am assuming you are receiving your incoming facsimile using SIP, if so I would suggest you have only one reference to t38pt_udptl in that peers configuration and set it to no. Depending on whether the peer is dedicated to receiving facsimiles I would suggest you also include in your peer's configuration faxdetect=no otherwise if this is an Audio/FAX line I would suggest you set it to faxdetect=cng. Once you have this working but really want to use T.38 then you will need to apply the T.38 Gateway patch to your 1.8.5.0 build, see https://issues.asterisk.org/view.php?id=13405 . Changes you will need to make to your SIP peer is to set t38pt_udptl=yes and in your dial plan before the Dial() enable the gateway with Set(FAXOPT(t38gateway)=yes). Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5
someone can help me to solve this problem? thanks -- From: Alessio ales...@asistar.it Sent: Friday, September 02, 2011 5:10 PM To: Lee Howard fax...@howardsilvan.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +asterisk1.8.5 1: from the phone i called the fax-server 2: from external fax i tried to send a fax to fax-server the results: 1: from the phone ( I hear sound of fax ) __ in asterisk CLI i have something like this: == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [06456789@PRINCIPALE:1] Goto(SIP/06456789-0003, IncomingFAX,06456789,1) in new stack -- Goto (IncomingFAX,06456789,1) -- Executing [06456789@IncomingFAX:1] Verbose(SIP/06456789-0003, * Call from Eutelia **) in new stack * Call from Eutelia ** -- Executing [06456789@IncomingFAX:2] Dial(SIP/06456789-0003, IAX2/iaxmodem) in new stack -- Call accepted by 127.0.0.1 (format ulaw) -- Called IAX2/iaxmodem -- Format for call is ulaw -- IAX2/iaxmodem-640 is ringing -- IAX2/iaxmodem-640 answered SIP/06456789-0003 -- Executing [h@IncomingFAX:1] Gosub(SIP/06456789-0003, riaggancia,~~s~~,1) in new stack -- Executing [~~s~~@riaggancia:1] Verbose(SIP/06456789-0003, ** Hungup **) in new stack ** Aggangio semplicemente la chiamata ** -- Executing [~~s~~@riaggancia:2] Hangup(SIP/06456789-0003, ) in new stack == Spawn extension (riaggancia, ~~s~~, 2) exited non-zero on 'SIP/06456789-0003' -- Hungup 'IAX2/iaxmodem-640' == Spawn extension (IncomingFAX, 06456789, 2) exited non-zero on 'SIP/06456789-0003' _ 2: from external fax __ in asterisk CLI i have something like this: ___ == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [06456789@PRINCIPALE:1] Goto(SIP/06456789-0003, IncomingFAX,06456789,1) in new stack -- Goto (IncomingFAX,06456789,1) -- Executing [06456789@IncomingFAX:1] Verbose(SIP/06456789-0003, * Call from Eutelia **) in new stack * Call from Eutelia ** -- Executing [06456789@IncomingFAX:2] Dial(SIP/06456789-0003, IAX2/iaxmodem) in new stack -- Hungup 'IAX2/iaxmodem-2218' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/0465940394-0002' status is 'CHANUNAVAIL' -- From: Lee Howard fax...@howardsilvan.com Sent: Friday, September 02, 2011 4:33 PM To: Alessio ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 The error happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. Alessio wrote: If I install asterisk i have the same problem. can anyone help me? thanks -- From: Lee Howard fax...@howardsilvan.com Sent: Thursday, September 01, 2011 6:29 PM To: ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore
Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
Thanks for the reply! i'll try to install asterisk on the same machine -- From: Lee Howard fax...@howardsilvan.com Sent: Thursday, September 01, 2011 6:29 PM To: ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
If I install asterisk i have the same problem. can anyone help me? thanks -- From: Lee Howard fax...@howardsilvan.com Sent: Thursday, September 01, 2011 6:29 PM To: ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
1: from the phone i called the fax-server 2: from external fax i tried to send a fax to fax-server the results: 1: from the phone ( I hear sound of fax ) __ in asterisk CLI i have something like this: == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [06456789@PRINCIPALE:1] Goto(SIP/06456789-0003, IncomingFAX,06456789,1) in new stack -- Goto (IncomingFAX,06456789,1) -- Executing [06456789@IncomingFAX:1] Verbose(SIP/06456789-0003, * Call from Eutelia **) in new stack * Call from Eutelia ** -- Executing [06456789@IncomingFAX:2] Dial(SIP/06456789-0003, IAX2/iaxmodem) in new stack -- Call accepted by 127.0.0.1 (format ulaw) -- Called IAX2/iaxmodem -- Format for call is ulaw -- IAX2/iaxmodem-640 is ringing -- IAX2/iaxmodem-640 answered SIP/06456789-0003 -- Executing [h@IncomingFAX:1] Gosub(SIP/06456789-0003, riaggancia,~~s~~,1) in new stack -- Executing [~~s~~@riaggancia:1] Verbose(SIP/06456789-0003, ** Hungup **) in new stack ** Aggangio semplicemente la chiamata ** -- Executing [~~s~~@riaggancia:2] Hangup(SIP/06456789-0003, ) in new stack == Spawn extension (riaggancia, ~~s~~, 2) exited non-zero on 'SIP/06456789-0003' -- Hungup 'IAX2/iaxmodem-640' == Spawn extension (IncomingFAX, 06456789, 2) exited non-zero on 'SIP/06456789-0003' _ 2: from external fax __ in asterisk CLI i have something like this: ___ == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [06456789@PRINCIPALE:1] Goto(SIP/06456789-0003, IncomingFAX,06456789,1) in new stack -- Goto (IncomingFAX,06456789,1) -- Executing [06456789@IncomingFAX:1] Verbose(SIP/06456789-0003, * Call from Eutelia **) in new stack * Call from Eutelia ** -- Executing [06456789@IncomingFAX:2] Dial(SIP/06456789-0003, IAX2/iaxmodem) in new stack -- Hungup 'IAX2/iaxmodem-2218' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/0465940394-0002' status is 'CHANUNAVAIL' -- From: Lee Howard fax...@howardsilvan.com Sent: Friday, September 02, 2011 4:33 PM To: Alessio ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 The error happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. Alessio wrote: If I install asterisk i have the same problem. can anyone help me? thanks -- From: Lee Howard fax...@howardsilvan.com Sent: Thursday, September 01, 2011 6:29 PM To: ales...@asistar.it Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5 Alessio wrote: I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. . Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING] Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2 Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without successful handshake Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r] Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK] It happens so quickly that I would suspect that it has to do with fax detection within Asterisk re-routing the call to a different place. Watch the CLI when a fax call comes in and see what happens there. However, let me say now that your setup that you describe strings the IAX2 channels out over your LAN which is no guarantee that there won't be jitter to cause you other problems. Normally iaxmodem (and probably therefore HylaFAX) should run on the same system as Asterisk. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update
[asterisk-users] from asterisk 1.6 to 1.8 - sip trunk unreachable
Hi! I recently upgraded Asterisk from version 1.6.2 to 1.8.5 Now about every 10 minutes all SIP TRUNKS becomes UNRECHABLE for a few seconds or minutes after become LAGGED and later become OK. I have no idea of the cause of this problem. With the version 1.6.2 all runs perfectly. I can't say more because I have no idea where the problem is. Any suggestions? thanks-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with hylafax + iaxmodem + asterisk1.8.5
Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other is the
server with hylafax and iaxmodem installed.
In Asterisk I set up an IAX trunk in this way:
___
iax.conf
[iaxmodem]
type=friend
context=outgoing-fax
disallow=all
allow=ulaw
username=iaxmodem
secret=password
qualify=yes
notransfer=yes
host=dynamic
requirecalltoken=no
callerid=Fax 06456789
t38pt_udptl=yes
___
In asterisk CLI when I write IAX2 show peers I read that the device is
reachable:
iaxmodem/iaxmod 10.0.1.202 (D) 255.255.255.255 4570 OK (3 ms)
In the end I put the configuration Hylafax and Iaxmodem.
I've created a context in Asterisk for incoming fax:
context IncomingFax {
_. = {
Dial(IAX2/iaxmodem);
};
h = {
riaggancia();
}
};
the call comes, the modem answers but does not receive any faxes.
I give you also logs /var/log/syslog and xferfaxlog
Thanks for your patience.
**
* config ttyIAX in /etc/iaxmodem/ttyIAX
**
device /dev/ttyIAX
owner uucp:uucp
mode 660
port 4570
refresh 300
server 10.0.1.204 // this is asterisk 1.8.5
peername iaxmodem
secret password
cidname FAXServer
cidnumber 0123456789
codec slinear
**
* config.ttyIAX in /var/spool/hylafax/etc/config.ttyIAX
**
CountryCode: 39
AreaCode: 06
FAXNumber: +39.06.456789
LongDistancePrefix: 0
InternationalPrefix: 00
DialStringRules: etc/dialrules
ServerTracing: 0xFFF
SessionTracing: 0xFFF
RecvFileMode: 0600
LogFileMode: 0600
DeviceMode: 0600
RingsBeforeAnswer: 1
SpeakerVolume: off
GettyArgs: -h %l dx_%s
LocalIdentifier: IAXmodem
TagLineFont: etc/lutRS18.pcf
TagLineFormat: Ricevuto da %%l|%c|Pagina %%P di %%T
MaxRecvPages: 200
ModemType: Class1 # use this to supply a hint
Class1AdaptRecvCmd: AT+FAR=1
Class1TMConnectDelay: 400 # counteract quick CONNECT response
ModemResetCmds: AT+VCID=1 # enables CallID display
PagerTTYParity: none
CallIDPattern: NMBR=
CallIDPattern: NAME=
CallIDPattern: ANID=
CallIDPattern: NDID=
***
xferfaxlog
**
09/01/11 17:13 CALL 00013 ttyIAX fax +39.06.456789 0 0 0:00:01
0:00:01 Ring detected without successful handshake NONE::s
***
/var/log/syslog
**
STATE CHANGE: RUNNING - LISTENING
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [9:DATE=0901]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [9:TIME=1650]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [5:NAME=]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [15:NMBR=0461829011]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [11:ANID=NONE]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [13:USER=iaxmodem]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [11:PASS=NONE]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [11:CDID=NONE]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [6:NDID=s]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 06654321
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 2
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 3 NONE
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 4 s
Sep 1 16:50:11 FAXServer FaxGetty[6225]: STATE CHANGE: LISTENING - ANSWERING
Sep 1 16:50:12 FAXServer FaxGetty[6225]: ANSWER: Ring detected without
successful handshake
Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [5:ATH0\r]
Sep 1 16:50:12 FAXServer FaxGetty[6225]: -- [2:OK]
Sep 1 16:50:12 FAXServer FaxGetty[6225]: MODEM set DTR OFF
Sep 1 16:50:12 FAXServer FaxGetty[6225]: MODEM set baud rate: 0 baud (flow
control unchanged)
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set DTR OFF
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set baud rate: 0 baud (flow
control unchanged)
Sep 1 16:50:13 FAXServer FaxGetty[6225]: DELAY 75 ms
Sep 1 16:50:13 FAXServer FaxGetty[6225]: MODEM set DTR ON
Sep 1 16:50:13 FAXServer FaxGetty[6225]: DELAY 2600 ms
Sep 1 16:50:17 FAXServer FaxGetty[6225]: MODEM set baud rate: 19200 baud,
input flow XON/XOFF, output flow XON/XOFF
Sep 1 16:50:17 FAXServer FaxGetty[6225]: DELAY 10 ms
Sep 1 16:50:17 FAXServer FaxGetty[6225]: MODEM flush i/o
Sep 1 16:50:17 FAXServer FaxGetty[6225]: -- [4:ATZ\r]
Sep 1 16:50:17 FAXServer FaxGetty[6225]: -- [2:OK]
Sep 1 16:50:17 FAXServer FaxGetty[6225]: DELAY 3000 ms
Sep 1 16:50:17 FAXServer HylaFAX[6247]: checkHostIdentity(localhost)
Sep 1 16:50:17 FAXServer HylaFAX[6247]: Parsing hostPort(): EPRT
Sep 1
[asterisk-users] call forwarding number from outside.
Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding number from outside.
Thanks for the reply! I've tried and works, but isn't possible with the transfer assisted? thanks From: Mike Sent: Friday, July 29, 2011 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] call forwarding number from outside. That`s the normal behavior of assisted transfers. Try a blind/non-assisted transfer, that should show the original callerid. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding number from outside. Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not the number 01234567890. How can you solve this problem? Thanks and sorry for my English -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding number from outside.
So I can't do anything? -- From: Kevin P. Fleming kpflem...@digium.com Sent: Friday, July 29, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding number from outside.
ok I'll do it Monday, and how you handle it with the version 1.10? Thanks -- From: Danny Nicholas da...@debsinc.com Sent: Friday, July 29, 2011 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. Upgrade to 1.8/10.0 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 10:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. So I can't do anything? -- From: Kevin P. Fleming kpflem...@digium.com Sent: Friday, July 29, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call forwarding number from outside. On 07/29/2011 10:41 AM, Danny Nicholas wrote: snip Couple of questions - This magic trick is contained in app_dial? Functionality is inherent to 1.8/10.X structure so we can't re-invent this in our old 1.4/1.6 installs? No, it's core functionality, implemented in the channel drivers and using control frames that pass through bridges. It would be a large amount of effort to implement it again in 1.4/1.6. It extends well beyond simple dialing, as it can receive updates across external protocols and pass them along, it handles call redirection, and various other features. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup(${EXTEN:2}); not works from outside
Hi!
I'm using ael language and I need to pick up a call from outside to an internal
number.
for example:
i'm 120
the phone 100 rings, it's a call from outside.
now I pick up the call with: *8100
and I would expect to answer the call but the response is Declined
the Puckup code is below:
_*8X! = {
Pickup(${EXTEN:2});
Hangup();
}
the problem is if an another number ( 130) calls the phone 120 and I pick up
with *8100, this works!
I respond to the caller 130.
I tried this code below:
_*8X! = {
SET(GLOBAL(PICKUPMARK)=${EXTEN:2});
Pickup(${EXTEN:2}@PICKUPMARK);
}
but if I type the number *8101 or *8104 or *8103 I always answer the call for
the number 100.
I hope I was clear
I'm sorry for my english.
Thanks--
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Re: [asterisk-users] Pickup(${EXTEN:2}); not works from outside
I think I have solved with the following code:
_*8X! = {
PickUpChan(SIP/${EXTEN:2});
Hangup();
}
thanks
From: Alessio
Sent: Friday, July 22, 2011 11:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Pickup(${EXTEN:2}); not works from outside
Hi!
I'm using ael language and I need to pick up a call from outside to an internal
number.
for example:
i'm 120
the phone 100 rings, it's a call from outside.
now I pick up the call with: *8100
and I would expect to answer the call but the response is Declined
the Puckup code is below:
_*8X! = {
Pickup(${EXTEN:2});
Hangup();
}
the problem is if an another number ( 130) calls the phone 120 and I pick up
with *8100, this works!
I respond to the caller 130.
I tried this code below:
_*8X! = {
SET(GLOBAL(PICKUPMARK)=${EXTEN:2});
Pickup(${EXTEN:2}@PICKUPMARK);
}
but if I type the number *8101 or *8104 or *8103 I always answer the call for
the number 100.
I hope I was clear
I'm sorry for my english.
Thanks
--
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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Hi, Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. Just my 2 cents I have more than 70 running servers installed with 1.2, we also built our custom interface around it, our custom linux/asterisk distro has been polished over the years and now finally we are earning the profit of all the work we did in the past. We just decided to open a new project with 1.4, but it will take us more than one year, i think, to release the first usable version. So, in the end, my opinion is that is just a matter of time. Hope it helps, have a nice Christmas everyone! -- I migliori saluti,Scrivi a: Alessio[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3 PCI slot with exclusive IRQ ? please advice!
Hi, I'm looking for an hardware platform for an * installation that should have at least 3 PCI slot with no irq sharing whatsoever. Hardware raid 1 with hot swap is a premium, but not mandatory ... What would you choose? compaq/hp ? Dell ? Ibm ? Tnx for any advice on this matter! -- I migliori saluti, Scrivi a: Alessio [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio from IVR of Cisco Call Manager
Hi all,we trying to integrate a Cisco Call Manager with the Asterisk PBX.We are running this configuration: the Call Manager is connected via SIP to Asterisk as a peer, and handles on its own a number of telephones. When the Call Manager receive a call the its own IVR starts the playback but no audio is heard to the caller. I also noticed the RTP stream in this stage is not up, and the connection is still in ringing state on the Asterisk side. Then after a while, the Cisco IVR transfers the call to an telephone, where picking up the conversation is heard. Any ideas or suggestions?Regards,Alessio ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN configuration
Hello Andrea, Thursday, July 6, 2006, 3:19:13 PM, you wrote: AS Hello everyone, AS I'm trying to set up an Asterisk machine with a quad-port BRI AS Junghanns card, and I want to use the mISDN drivers. Hi andrea, best way to do it is to download the install-misdn mqueue from beronet web site http://www.beronet.com/index.php?option=com_remositoryItemid=38func=selectfoldercat=1lang=en just untar it and do make install, it will download and compile all needed files for misdn - chan_misdn. If you need further assistance you can contact me (in italian also) al alessio AT interconnessioni DOT it Regards! AS I'm having some trouble configuring it: do I need to use CAPI drivers? AS I haven't found good links, could you please provide some info? AS Thanks in advance, -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use ringall with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that nobody wants that (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a count us initiative :)Alessio FocardiOn 6/29/06, Aaron Paxson [EMAIL PROTECTED] wrote: I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list? ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
Will you (or anyone else) be able to code this proposed circular or linear (what sounds more appropriate?) strategy and submit it for inclusion in HEAD ?Should be pretty easy, unfortunately I have very few programming skills. Regards !P.S.here is a snippet from the wiki, whatever it means ! :)roundrobin mode remembers the last agent it _started_ with for a new call, and starts with the next agent in the list. If you have three agents, the first call will go to agent 1-2-3, the next call will go to 2-3-1, the next call will go to 3-2-1, etc. rrmemory mode remembers the last agent it tried to _call_, regardless of who it started with, so that the next call will go the agent after the last one who answered. If you have three agents and the first call rings 1-2 (and is answered), then the next call will ring 3-1 (and is answered), then the next call will ring 2-3-1, etc. For the first call, if agent 2 answered it in roundrobin mode, they would still be the first agent for the next call, but rrmemory mode will move past them. On 6/29/06, Aaron Paxson [EMAIL PROTECTED] wrote: The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - From: Alessio Focardi To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, June 29, 2006 1:31 PM Subject: Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use ringall with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that nobody wants that (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a count us initiative :)Alessio Focardi On 6/29/06, Aaron Paxson [EMAIL PROTECTED] wrote: I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list?___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WIFI sip phone
Hi folks!Based upon your experience on the field what wifi sip phone would youreccomend ?A customer asked for a wireless * install and I'm looking for advice, tnxAlessio Focardi[[*] - Interconnessioni Italy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Hi, Ciao I have a bunch ( 30) 320's connected to HP switches, fw version 6.0.4 They work great but occasionally they where signalling the warning network cable disconnected. Monday, May 29, 2006, 10:38:09 AM, you wrote: TC Guido Hecken wrote: I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users TC I have moved to 3com switches,but the Snom 320 still locks up, and also TC I don't think it's reasonable to force customers to buy 3com just TC because Snom firmware sucks. TC ___ TC --Bandwidth and Colocation provided by Easynews.com -- TC Asterisk-Users mailing list TC To UNSUBSCRIBE or update options visit: TChttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?
Hello Cosmin, Friday, May 12, 2006, 10:45:05 AM, you wrote: CP Hello everyone. CP I've got a HFC ISDN card that I'm using with chan_misdn and it basically CP behaves like crap. Echo is waaay worst then echo I get TDM400 card, CP sound is choppy (there other side is allays complaining about sound CP interruptions) and to top it all it detects fake DTMF's all the time. Try this settings for echo cancel: in my setup they work wery well (most of the times) [g1] echocancel=256 echotraining=no jitterbuffer=4000 jitterbuffer_upper_threshold=0 -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help in asterisk fax
On 5/2/06, Gidean Chan [EMAIL PROTECTED] wrote: Can anyone tell me how to make it work? I have asterisk 1.10.006 and hylafax in the same linux server. 2 x100p on PCI slots connected with 2 PSTN lines.In my opinion you have two options:1) setup iaxmodem for hylafax and use asterisk as pbx and hylafax for faxing; you can also host the both on the same server 2) install app_txfax and app_rxfax for asterisk and use them to send and receive faxIf you have more experience in hlyfax than asterisk probably you should go with the first solutionHope it helps! I was using hylafax on one line with an external modem before. Now I have already removed the external modem and want to use asterisk to receive fax. (fax to email, no need email to fax). Please explain in detail as I dont even know how to install or complie. Thank you very much!! Gidean Chan ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hinting a conference room
Hi there!I was asked to set up a led on a snom phone monitoring a conference room (lit when someone is in conference).I know that there is a patch for hinting parking lots, anyone has made something similiar for conferences ? Tnx for the support!P.S.What about monitoring a global var ?It would be absolutely great variable=0 led off, 1 led on, 2 led blink ... Alessio Focardi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.
On 4/4/06, Pimjai Wesnarat [EMAIL PROTECTED] wrote: I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfullybut I have problems with some fax machine so I wanted to try usingHylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. Mysystem looks like this:This is my working iax.conf hylafax section[iaxmodem1]type=friendsecret=blahblah host=dynamiccontext=lcr_defaultdisallow=allallow=slinearand this is the iaxmodem1 config for iaxmodemdevice /dev/ttyIAXowner uucp:uucpmode 660port 4570 refresh 60server 172.16.199.196peername iaxmodem1secret blahblahcidname fax1cidnumber codec slinear Hope it helps! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over 2 bridged TE110P channels
On 4/4/06, Remco Barende [EMAIL PROTECTED] wrote: I suspect that in your case the fax channels are not natively bridged. I'mnot sure whether native bridging will work if you are using 2 cards.Tnx to all who replied to me, I'm posting a reply I received from Digium support for reference to other having the same problem: Officially, fax communications is not supported, because the precisionof the timing and jitter is not sufficient for clean enough transmissionof the fax signalling to offer 100% reliable performance. That being said, faxing generally works over bridged TDM connections.The problem you are likely having is one or both of the following:1. The timing sync of the cards is not going to work as desired: You are taking timing from the telco (assumably) on one card, and in order forthe fax bridge to work properly, you'd have to re-send that timingsignal out the other card. The timing coming out of the other card (thepri_net span) is not synced to the other span -- it is being generated by the internal clock on the card itself. Due to this, the timing syncis not making it from the telco to the other PBX. It is possible to syncspans across cards only with the 2 and 4-span cards using a timing cable between them. It is also possible to sync timing if you had a singledual-span card servicing both the E1s instead.2. Less likely: the bridge across the spans is traversing the PCI bus,and system resource contention might be tainting the signal, introducing jitter, or possibly dropping frames.You might want to check with Digium support to verify Let me know the result.Cheers!RemcoOn Tue, 4 Apr 2006, Alessio Focardi wrote: Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is simply redialed in the pbx every call from pbx is simply redialed to the line no answer is done All is working great with voice, but faxing often results in error, both receiving and sending. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over 2 bridged TE110P channels
Hi,I have an asterisk installation with 2 E1 cardsSoftware version isAsterisk 1.2.6Libpri 1.2.2Zaptel 1.2.5I'm having problem with fax transmission, let me explain better mysetup:
My fist TE110P E1 card is connected to the telco linethe second TE110P E1 one to an Nexspan PBXso the server is basically sitting between the line, and the pbx.every call coming from the line is simply redialed in the pbx
every call from pbx is simply redialed to the lineno answer is doneAll is working great with voice, but faxing often results in error, bothreceiving and sending.I have disabled echo cancel, and also checked for interrupts problems
and other common misconfiguration problems.Would someone please help me sort this out ?I'm suspecting sync problems ...Tnx for any help!Following are some debug and config files
zaptel.confloadzone = itdefaultzone = itspan=1,1,0,ccs,hdb3,crc4bchan=1-15dchan=16bchan=17-31span=2,0,0,ccs,hdb3,crc4bchan=32-46dchan=47bchan=48-62
zapata.conf[channels]switchtype = euroisdn;linesignalling=pri_cpepridialplan=unknownswitchtype=euroisdnpriindication = outofbandechocancel=nooverlapdial=yes
immediate=nonationalprefix=internationalprefix=resetinterval=300context=pri1group=1channel = 1-15channel = 17-31;pbxsignalling=pri_netpridialplan=internationalswitchtype=euroisdn
priindication=outofbandechocancel=nooverlapdial=yesimmediate=nonationalprefix=internationalprefix=resetinterval=300context=pri2group=2channel = 32-46channel = 48-62
pri1 contextexten=_X.,1,Dial(Zap/g2/${EXTEN}||j)exten=_X.,2,Congestion()exten=_X.,102,Busy()pri2 contextexten=_X.,1,Dial(Zap/g1/${EXTEN}||j)exten=_X.,2,Congestion()
exten=_X.,102,Busy()cat /proc/interrupts CPU0 0: 1114420235 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 1114083499 XT-PIC t1xxp
8: 1 XT-PIC rtc 9: 0 XT-PIC acpi10: 2531734 XT-PIC eth012: 1114121836 XT-PIC t1xxp14: 306435 XT-PIC ide0NMI: 0
lspci -v00:00.0 Host bridge: Silicon Integrated Systems [SiS] SiS645 Host Memory AGP Controller (rev 01) Flags: bus master, medium devsel, latency 32 Memory at e000 (32-bit, non-prefetchable) [size=64M]
Capabilities: [c0] AGP version 2.000:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI bridge (AGP) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 64
Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: dde0-dfef Prefetchable memory behind bridge: d9c0-ddcf00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS961 [MuTIOL Media IO]
Flags: bus master, medium devsel, latency 000:02.1 SMBus: Silicon Integrated Systems [SiS] SiS961/2 SMBus Controller Flags: medium devsel I/O ports at 0c00 [size=32]00:02.5
IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev d0) (prog-if 80 [Master]) Subsystem: Silicon Integrated Systems [SiS] SiS5513 EIDE Controller (A,B step) Flags: bus master, fast devsel, latency 128
I/O ports at ff00 [size=16]00:03.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI Fast Ethernet (rev 90) Subsystem: Silicon Integrated Systems [SiS] SiS900 10/100 Ethernet Adapter
Flags: bus master, medium devsel, latency 64, IRQ 10 I/O ports at dc00 [size=256] Memory at dfffc000 (32-bit, non-prefetchable) [size=4K] Expansion ROM at dffa [disabled] [size=128K]
Capabilities: [40] Power Management version 200:08.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 16b8 (rev 01) Subsystem: Cologne Chip Designs GmbH: Unknown device b562 Flags: medium devsel, IRQ 11
I/O ports at d800 [size=8] Memory at d000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 200:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
Subsystem: Unknown device 6159:0001 Flags: bus master, medium devsel, latency 64, IRQ 5 I/O ports at d400 [size=256] Memory at dfffe000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 200:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 6159:0001 Flags: bus master, medium devsel, latency 64, IRQ 12
I/O ports at d000 [size=256] Memory at dfffd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 201:00.0 VGA compatible controller: nVidia Corporation NV5 [RIVA TNT2/TNT2 Pro] (rev 11) (prog-if 00 [VGA])
Subsystem: Diamond Multimedia Systems Viper V770 Ultra Flags: bus master, 66Mhz, medium devsel, latency 64 Memory at de00 (32-bit, non-prefetchable) [size=16M] Memory at da00 (32-bit, prefetchable) [size=32M]
Expansion ROM at dfef [disabled] [size=64K] Capabilities: [60] Power Management version 1 Capabilities: [44] AGP version 2.0
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Re: [Asterisk-Users] beronet bristuff
Ciao Francesco, in data mercoledì 22 marzo 2006, alle ore 13.04, hai scritto: On Beronet installation manual I read that Beronet and Junghanns cardsare identical in their construction but Junghanns made bristuff so that only their cards can work with their drivers. In the same document and googling around I found that bristuff source files must be patched to recognize other cards. Anyone has experienced with this? The alternative would be using mISDN, but Im not sure its as fine as bristuff. We had for testing purposes, look for pci id of card in source code, I'm not telling more about that since it may seem rude. Give misdn a test, it works better every day! Saluti, mandami un messaggio privato ad alessio AT interconnessioni PUNTO it se vuoi continuare la discussione. -- I migliori saluti,Scrivi a: Alessio[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting a PRI failure from dialplan
Hi,
I would like to know if there is a way to detect the status of a span
prior of sendig a call across it from the dialplan.
I was asked to set up an * server with 2 spans connected to the telco and use
the second as
failover for the first.
I checked that dialing a failed span (for instance: cable disconnected or
no line) results in congestion for ${DIALSTATUS}, but message is too
generic for my use.
Any suggestion will be greatly appreciated, tnx!
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Pls. explain what happens...
Hello Mauro, Tuesday, December 27, 2005, 9:26:54 AM, you wrote: MZ Hi everybody, MZ can anybody explain one thing: say we have 2 SIP phones(or H323) and one MZ Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk and MZ phon3 answers: is the real conversation streaming thru the * box, or it's MZ going straigth from one phone to the other? FOr what I know, It depends: if you have reinvite set to yes in sip.conf, and there is no need of format transcoding, the audio stream goes directly from phone to phone. To check that just unplug the ethernet cable from the asterisk server while having a conversation between 2 phones: call should stay up. Hope it helps! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123
GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you also need to change GK musiconhold.conf from mode=quietmp3 to mode=files. Regarding this issue: anyone knows how to setup streaming music on hold (from webradios) with the new native syntax ? Previously I was using this as suggested by the wiki: radiowazee= mp3:/var/lib/asterisk/sounds/pbx/webradio,http://grace.fast-serv.com:9206/ where in the webradio dir there was just a dummy mp3 file I would like to reproduce this using native mp3 ... any idea ? Tnx ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hello richard, Wednesday, November 23, 2005, 4:54:54 PM, you wrote: rC Alessio, Sergio So an upgrade is of course necessary. rC i have upgraded the vigor. Bad news... i am not able rC to register the draytek anymore. But using a XLite on rC my pc behind the Vigor works now fine (no one way rC audio). rC however i have an other question. I saw you put for rC the bindaddr same thing like 192.168.0.3. Is that the rC ip addr from your Asterisk? Yes it is ... we are using this vigor # Model : Vigor2600V series annex A # Firmware Version : v2.5.5.3_I # Build Date/Time : Fri Dec 31 10:37:6.33 2004 Hope it helps! ___ --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 and DrayTek Vigor2600VGi
Hello richard, Wednesday, November 23, 2005, 10:46:03 AM, you wrote: rC Hi all, rC I'm trying to configure a remote user with a DrayTek rC 2600Vgi. The setup looks like this. rC [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I tried a similar setup some times ago and it was working, have you put the private ip address of the asterisk box in the vigor setup ? Can you ping the private address of the vigor from the asterisk box and viceversa ? Hope it helps ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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[2]: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
Hello richard, Wednesday, November 23, 2005, 12:34:33 PM, you wrote: rC Hi Alessio rC [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone] I tried a similar setup some times ago and it was working, have you put the private ip address of the asterisk box in the vigor setup ? Can you ping the private address of the vigor from the asterisk box and viceversa ? rC I am able to ping the private addr of the vigor from * rC and of couse viceversa. The vigor setup seems to be ok rC (vpn is up and *sip show peers* shows that the vigor rC is registred.). I can also call from and to Asterisk, rC so the signalisation is ok. I have only problem with rC RTP packets (one way audio) I'm having 10 peers over vpn vith 10 vigor in a customer setup, here is a sample of my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.3 ; Address to bind SIP channel to context = default ; Default context for incoming calls ;srvlookup = yes; Enable DNS SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ; IP QoS parameter, either keyword or value disallow=all ;allow=all ;allow=gsm allow=g729 ;allow=ilbc ;allow=ulaw ;allow=alaw [3200] ;LA SPEZIA 1 ;DRAYTEK VIGOR 2600 type=friend host=dynamic username=3200 secret=* canreinvite=yes context=sip qualify=yes I will also suggest to nail up the vpn connection from the vigor and upgrading the vigor firmware . Wish you luck! rC __ rC Yahoo! FareChase: Search multiple travel sites in one click. rC http://farechase.yahoo.com rC ___ rC --Bandwidth and Colocation sponsored by Easynews.com -- rC Asterisk-Users mailing list rC Asterisk-Users@lists.digium.com rC http://lists.digium.com/mailman/listinfo/asterisk-users rC To UNSUBSCRIBE or update options visit: rChttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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 1.2 Released!
Congratulations from Italy now back to work for 1.3 ! :) -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] newbie questions
Hello Hiu, Monday, November 7, 2005, 4:51:35 AM, you wrote: HYO i am pretty new to asterisk. hope to learn more. HYO i have this notice from the console. when i was doing the echo testing HYO by putting the context=default. then, i called out 600 to get the echo HYO test, i can hear the operator talking, but i cant really hear the playback. HYO i am trying to dig around from info from the log files. HYO what does it mean? HYO RFC3389 support incomplete. Turn off on client if possible HYO hope to help..thanks That means that you have to turn off silence suppression in your softphone (in xlite is named transmit silence). Hope it helps! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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 Addons linker's error....
Hello Mauro, Monday, November 7, 2005, 11:21:25 AM, you wrote: MZ Hi Everybody, MZ I have spent last 3 days in trying to compile Addons. 1.0.9 MZ I have succeded in reaching linker's phase. The linker is unable to find MZ both z lib and mysqlclient lib. MZ Many tests in trying to change z to zlib.so(available on /usr/lib) and MZ libmysqlclient.so (existing on /usr(lib/mysql). MZ Still the problem is there! (going nuts...) I'm using Fedora, normally I use ldconfig after installing libraries. http://www.die.net/doc/linux/man/man8/ldconfig.8.html Hope it helps! P.S. italiano ? :) -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] Called number (Destination Number)
Hello David, Friday, November 4, 2005, 11:27:51 AM, you wrote: DA Hi, DA I have E1 PRI, When I have an incoming call, how can I know the called DA number (or the destination number) before answer the call? DA My provider say that he send it. DA E1 PRI DA 900XX 9XXX -- Asterisk Maybe you have immediate=yes in zapata.conf and all calls are coming in to s extension. Try to set immediate=no in zapata.conf for the span: you should be able to see on the cli the called number. Then you will have to create the relative extensions in the incoming context ... just s will not work anymore. Hope it helps! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] REGEX() 1.2beta2
Hi, anyone has a working example of this new function ? that's all that I have found -= Info about function 'REGEX' =- [Syntax] REGEX(regular expression data) [Synopsis] Regular Expression: Returns 1 if data matches regular expression. [Description] Not available Tnx! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --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] Agent logout
Hi, is there an Agentlogout procedure opposite of the one we get with Agentlogin ? I tried simply having another agent log from the same extension, but when I try Show agents 10 (Alessio) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 51 (Giuliano) available at '[EMAIL PROTECTED]' (musiconhold is 'default') So another question could be: to who calls are counted if answered ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --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] Url dialing
jsss My suggestion would be the one-line eyeBeam phone under jsss development. Check out support.xten.com. I checked a multiline versionof eyebeam: no url opening within the phone call, using this syntax: Dial(sip/399|||http://www.google.it) Could it be that only IAX2 supports this ? Tnx! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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[2]: [Asterisk-Users] asterisk certification - thread hijack
I took the certification in Astricon Madrid, still I have to get any kind of proof/certificate. I contacted the testing company and they told me it was just a matter of time, so probably they are working on this probably those are just super rapid growing problems. Regards! s The original poster's statement about not even receiving any s proof thathe was certified is kind of amazing. s I wouldn't be too upset about it either because it is probably s anhonest mistake, but I would be firm on demanding that you get s what youpaid for. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] Date based context inclusion
Hi, I know that writing in the dialplan include = day|09:00-19:59|mon-fri|*|* day will be include monday TO friday What is needed to include day monday AND friday ? include = day|09:00-19:59|mon,fri|*|* does not work, but it was just my guess Tnx for any help -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --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[2]: [Asterisk-Users] Snom 360 and hints
Hello BJ, Thursday, September 1, 2005, 2:06:43 PM, you wrote: BW Issue #3644 has recently been committed to CVS-HEAD which BW allows for full device state notification via subscriptions for BW Snom 360 and other supporting phones w/o the need for additional BW patches. Leds are working using CVS head, let's hope the patch gets into 1.2! Have you got any idea on how to setup call pickup pressing the blinking button on snom phones? Tnx! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] Snom 360 and hints
Hello Paul, Thursday, September 1, 2005, 4:38:42 AM, you wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) Same here, it think it depends on hint status: when you make a call calling hint is set to 1, but called one stays 0. Correct behaviour should be put the hint of the caller to 1 (steady ligt) while calling put the hint of the called to X (blinking light, cant remember which state it is ) while phone is ringing, then to 1 if call is answered. Unfortunately I dont know how to accomplish this Regards! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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[2]: [Asterisk-Users] Re: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?
Hello Lars, Have you got kernel sources installed ? I think that are mandatory for Zaphfc. Regards Tuesday, August 23, 2005, 10:37:37 AM, you wrote: LD On 8/22/05, Massimo De Nadal [EMAIL PROTECTED] wrote: Forget RPM. LD I agree. First of all read: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+SuSE LD we did then download http://www.junghanns.net/downloads/bristuff-0.2.0-RC8n.tar.gz explode the tarball, read the file INSTALL and run install.sh LD we did, and install.sh fails because it cannot compile zaphfc... LD I think we'll change distro or something. LD Lars. LD ___ LD Asterisk-Users mailing list LD Asterisk-Users@lists.digium.com LD http://lists.digium.com/mailman/listinfo/asterisk-users LD To UNSUBSCRIBE or update options visit: LDhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] REGEX Function
Hi,
anyone can write down a working example of a regex fuction ?
I'm using this syntax
Gotoif($[${REGEX(/B/ | A)}=1]?20)
But function always return 1, even if I write
Gotoif($[${REGEX()}=1]?20)
Tnx for any help !
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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[Asterisk-Users] Roundrobin queue strategy broken ?
Hi there, this is my queues.conf, I'm using todays CVS: [599] joinempty = yes musiconhold = default strategy = roundrobin servicelevel = 60 wrapuptime = 0 maxlen = 0 timeout=15 announce-frequency = 15 member = SIP/381 member = SIP/300 At first call 381 rings, if you hang up and call again you get the 300 ringing ... this looks more rrmemory than roundrobin, there is something wrong in my setup maybe ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] REINVITE and Codecs
Hi, just a question: Let say I have 2 phones with G729 onboard, but no 729 licence for Asterisk. Preferred codec set up in phones is G729, followed by ULAW, in Asterisk I have allow=ULAW deny=ALL. When call is reinvited by Asterisk will the two phones use G729 between each other or they will stick to ULAW they used for first part of the call ? A quick test showed that they will use ULAW ... can I work around this or am I getting something wrong ? Tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail : Unable to create lock file: No such file or directory
Hi, I get this message after password request in voicemail app: Unable to create lock file: No such file or directory Anyone got a clue about fixing that problem ? I can't understand what directory or file we are talking about .. Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. - You get a call - You want to transfer it - You hit # - You are presented a tone - You dial the extension you want to transfer to Now the hard part - If you hang up prior of the other party has answered you get an unattended transfer if, for any reason the other party dont answer (busy, no answer, wrong extension etc) call should be bounced back to you - If you stay on the phone and the other party answers you talk to him, introduce the call then hitting # again will switch back and forth between the call you are tranfering and the transfer party if you hang up call is trasfered to the other party if the other party hangs up you get back to the original call Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. What do you think about this flow ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Adam, In my opinion there should be only one transfer function, let suppose it's called by #. AG Wrong, which other phone system have you used where every time you try AG and use some IVR that says Enter your xyz number followed by the # key AG and you end up being interrupted by asterisk to transfer the call ?? Well as you can see it was an example, actually you have to decide this mapping in features.conf, so what's the point ? Let say is *# or any other sequence :) Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. AG Nope, because if there are three parties: AG A - You AG B - Outside caller 1 AG C - Outside transfer party AG When you hangup, you don't want the other two legs to stay up, AG potentially forever depending on your hangup detection etc... I know what I want! :) Why not, I'm announcing a call, then going conference, then leaving because I already did my part, why the other 2 calls have to be disconnected ... because hangup detection works bad ? What do you think about this flow ? AG Any SIP phone (decent one) should have much more intuitive/instructive AG transfer process. All I'm asking is a native function that can be used regardless of the UA, if you got such functions integrated in the phone, better yet, is up to you to choose then. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATXFER discussion, what's your opinion ?
Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit transfer then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang immediately after the transfer sequence the call is just transfered, and if the other party is busy or does not answer the transfered call is bounced back to you again. That's how pbx's users are expecting call transfer to work, is there a way to reproduce this behavior in asterisk ? For what I can see it's not possible and you will have to select two codes, one for blind and one for attended tranfers What do you think about it ? -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Michael, Wednesday, July 20, 2005, 11:54:40 AM, you wrote: MP Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. MP SNIP MP I honestly think that transfers is one thing that Asterisk should MP improve a LOT to be able to stand up to even the most cheapo taiwanese MP no-name PBXs, which support attended transfers out of the box. That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe we can try to write down a sort of flow chart of a new transfer function and then set up a bounty, anyone else would like to join me ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco Call Manager
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk Restart after crash
Hello Chuck, Friday, April 22, 2005, 5:56:02 AM, you wrote: CS Does Asterisk restart itself if it crashes? If not is there a way to make CS linux do it? Have a look at safe_asterisk script it's included, I guess. CS ___ CS Asterisk-Users mailing list CS Asterisk-Users@lists.digium.com CS http://lists.digium.com/mailman/listinfo/asterisk-users CS To UNSUBSCRIBE or update options visit: CShttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcte11xp works only after cold reboot
Hi, my brand new wcte11xp works like a charme of first boot, then if I shutdown -r now the server is not detected at reboot (no such device after modprobe). Turning off the pc and cold restarting fixes the problem. Has someone experienced such behaviour before ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P install problems
Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] TDM400P install problems
Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] TDM400P install problems
Hello Dana, Friday, March 18, 2005, 3:40:21 PM, you wrote: DO Can you run dmesg after that command and tell us what the relevant output is? # modprobe zaptel modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file or directory # dmesg Zapata Telephony Interface Registered on major 196 # I have to say that there are 2 cards in this server, this is my zaptel.conf fxoks=32-35 loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 was running cvs-head, now running 1.0.6 It seems that when I call wcfxs wctdm is called instead. Any idea ? TNX ! DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[6]: [Asterisk-Users] TDM400P install problems
Hello Scott, Friday, March 18, 2005, 5:10:14 PM, you wrote: SG Try using module wctdm instead. That solved a lot of headaches for me. There is no wctdm module in zaptel-1.0.6.tar.gz . So why when I call wcfxs ... modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.10-1.770_FC3/misc/wctdm.ko': No such file or directory That does not look normal to me, I have built another kernel to try to make this behavior go away, still no luck Tnx anyway ... SG On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi SG [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:40:21 PM, you wrote: DO Can you run dmesg after that command and tell us what the relevant output is? # modprobe zaptel modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file or directory # dmesg Zapata Telephony Interface Registered on major 196 # I have to say that there are 2 cards in this server, this is my zaptel.conf fxoks=32-35 loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 was running cvs-head, now running 1.0.6 It seems that when I call wcfxs wctdm is called instead. Any idea ? TNX ! DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk-addons: libmysqlclient.so.14: cannot open shared object file
Hi, I have compiled asterisk-addons successfully, but when I put res_config_mysql.so in modules directory asterisk fails to load, here is the error: 7:29 WARNING[19097]: loader.c:301 __load_resource: libmysqlclient.so.14: cannot open shared object file: No such file or directory Feb 17 15:17:29 WARNING[19097]: loader.c:509 load_modules: Loading module res_config_mysql.so failed! libmysqlclient is present on the system, should I edit something to point * to the right directory for it or something like ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Problem with asterisk-addons:libmysqlclient.so.14: cannot open shared object file
MB The Makefile seems to check most places for mysql libraries but check it MB again to make sure. Also make sure your mysql lib path is in ld.so.config MB then rerun ldconfig. (Oh..do that before you do the above commands) That was the problem, tnx ! P.S. Any skill in realtime ? I'm struggling to get it working with the BRISTUFFED version of * -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff and Realtime
Hi, I would like to use Realtime extentions with a four bri card, the classic quodbri. Normally with that card I would use * bristuffed from Klaus-Peter Junghanns, but since that package is based on stable version there is no Realtime at all in it (I suppose). Did you knoww if someone has done a merger, or can help me in such task ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff and Realtime
Hi, I would like to use Realtime extentions with a four bri card, the classic quodbri. Normally with that card I would use * bristuffed from Klaus-Peter Junghanns, but since that package is based on stable version there is no Realtime at all in it (I suppose). Any idea, other than wait for realtime to begin stable ? :) Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[3]: [Asterisk-Users] REALTIME and VARIABLES
Hello Dominique,
Tuesday, January 18, 2005, 4:28:19 PM, you wrote:
D How are you using the variables in your realtime table?
D Note that you will need to specify variables with the following syntax:
D SIP/${ARG1}|${ARG2} and not SIP/${ARG1},${ARG2} as in extensions.conf
well the problem just shows if I use a variable as an extension, like
in the example
No realtime:
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
all ok!
With realtime:
[globals]
IPPHONES=_3XX
[sip]
switch=Realtime/sip
exten=${IPPHONES},1,Answer in the table
it does not work anymore,${IPPHONES} is not solved as _3XX.
If I change in table ${IPPHONES} with _3XX all returns normal.
So my conclusion is that variables can not be used as extensions in
realtime contexts, the actually work for all the other usual purposes
anyway.
--
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Alessiomailto:[EMAIL PROTECTED]
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[Asterisk-Users] Using a variable for EXTEN
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make a dialplan without fixed extension
numbers to change the entire dialplan according to the customer
requests: what exten number do you want for your IP Phones ? let me change
a variable and we are set!
It seems that this is not supported, am I getting somethig wrong in
the syntax? There is another way to accomplish that ?
Tnx!
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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Re[2]: [Asterisk-Users] Using a variable for EXTEN
Hello Dave,
Monday, January 17, 2005, 12:50:13 PM, you wrote:
DC On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make a dialplan without fixed extension
numbers to change the entire dialplan according to the customer
requests: what exten number do you want for your IP Phones ? let me change
a variable and we are set!
It seems that this is not supported, am I getting somethig wrong in
the syntax? There is another way to accomplish that ?
DC Works like that for me (tm). If it's not working look carefully at all
DC your config files.
I'm doing this using realtime, so really the dialplan is
[globals]
IPPHONES=_3XX
[sip]
switch=Realtime/sip
Then in the db I have the extentions using the variable name.
Someone ever tried this with realtime ?
--
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Alessiomailto:[EMAIL PROTECTED]
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[Asterisk-Users] REALTIME and VARIABLES
Hi,
I'm having some problem with realtime:
let's say I have a dialplan like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
A call from ip phone 300 comes in, and it's been answered.
Then I switch the sip context to realtime, putting the exten in the
db and using the variable name for this as in the file version.
[globals]
IPPHONES=_3XX
[sip]
switch=Realtime/sip
Calling does not work anymore, extension is not found.
So it seems me that with realtime we cant'use variables as extensions
for an easyer manteniance of the dialplan.
Am I getting it all wrong ?
Tnx for any suggestion!
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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Re[2]: [Asterisk-Users] REALTIME and VARIABLES
Hello Matthew, Monday, January 17, 2005, 4:34:16 PM, you wrote: So it seems me that with realtime we cant'use variables as extensions for an easyer manteniance of the dialplan. MB Doesn't RealTime itself make for easier maintenance of extensions since MB its database driven? So this is not a bug, it's a feature! :) Seriously, anyone verified my problem and it's willing to share a solution if there is any ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime and include
Hi, I'm testing realtime right now, it does not seem to me that realtime contexts can be included in normal context, like this [sip] include=sip-dial exten=i,1,Hangup [sip-dial] switch=Realtime/sip-dial Am I getting it wrong ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] More Zaptel problems
i Channel 01: FXS Kewlstart (Default) (Slaves: 01) i Channel 02: FXS Kewlstart (Default) (Slaves: 02) i 2 channels configured. i But when I try to start asterisk, I get a box crash just after parsing i musiconhold. i Any clue? * hangs or crashes ? And if it crashes: have you got any message on screen ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mysql and Voicemail
Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then when I start * I have no results, show voicemail users -- There are no voicemail users currently defined also if I try to check against a box with MailboxExists it does not result created Any idea of what I'm getting wrong ? tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel ISDN BRI settings for The Netherlands KPN
Hello Remco, Tuesday, December 28, 2004, 10:53:46 AM, you wrote: RB Does anybody know the correct settings for this? I will not have internet RB access there which makes it harder to google around on location. RB switchtype = euroisdn RB is pretty obvious but what about these settings: RB signalling = bri_cpe_ptmp RB ; p2p TE mode RB ;signalling = bri_cpe RB ; p2mp NT mode RB ;signalling = bri_net_ptmp RB ; p2p NT mode RB ;signalling = bri_net RB pridialplan=local RB prilocaldialplan=local RB ; trust user provided callerid (clip no screening)? RB pritrustusercid = yes RB immediate=yes To my experience you will need bri_cpe as signaling (point2point), immediate=no (if you have more than one numer on the ISDN), pritrustusercid = no. Regarding the pridialplan and prilocaldialplan I suggest to make some experiment there. Hope it helps ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Mysql and Voicemail
Hello Matthew, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database Tuesday, December 28, 2004, 3:33:35 PM, you wrote: MB What version of Asterisk? CVS-HEAD I think I understood the problem: things are changed since the wiki page :) If I understood correctly now we have the possibility to externalize some of the confs using the realtime application. I have put voicemail and sip friends in sql tables that way, and it's working (more or less). Can someone more skilled than me describe what are the significant changes this addon have brought in * and what are the differences between realtime and config in sql ? Tnx for the support ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] Mysql and Voicemail
Hello Matthew, Tuesday, December 28, 2004, 3:56:26 PM, you wrote: MB RealTime is not an addon. Its written into the core code of asterisk. :) MB The advantage is so that you can manage most of asterisk via database MB instead of flatfiles. If you want to continue to use flatfiles, then do so. MB RealTime does NOT force you to use itself. Sure, I'm testing it right now ... looks VERY nice, writing a gui or automating some common task now looks a lot easier! The link for anyone interested: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disabling ! command
Hi, since I run asterisk as root with a CLI open on TTY12 I was wondering if the ! (shell) command can be disabled from the config, for safety reasons it seems me usefully. Tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Disabling ! command
Hello Justin, Friday, December 17, 2004, 3:43:12 PM, you wrote: JC you could comment that portion out and rebuild? You are right, I will do like this (well at first I have to understand where the comment has to be put) ... just wondering if maybe we can suggest a new option in the config for the purpose. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Recomended ISDN for Asterisk ?
Hello HBK, Monday, December 6, 2004, 12:11:45 PM, you wrote: H Hi H I have installed the http://asteriskathome.sourceforge.net/ with a H Digium card with no problems, very good ! H Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. H I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to H install Capi ! H Please suggest best and easiest approach ? I'm pretty satisfied with bristuff package from http://www.junghanns.net/asterisk/ It downloads, patch and compile asterisk for HFC cards, also TE mode is supported. Good luck! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voice in Asterisk with BRI ISDN Any proper working configurations yet?
Hello Christiaan, Thursday, November 18, 2004, 2:40:00 PM, you wrote: CB Hi all CB CB If been working a while now trying to interface Asteriskswith CB BRI ISDN. Ive tried various drivers without any success. CB Imrunning a HFC passive ISDN board in 2.6.5 kernel. are you using the bristuffed version of asterisk ? http://www.junghanns.net/asterisk/ Exactly what is the problem you are experiencing ? Regards ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom and Stun
Hi, I'm testing a snom 190 behind nat, using an external stun server. Problem is that the phone first register itself with the internal ip address, then checks for stun and registers again with the correct ip address. After the registration timeout the phone register just once with the correct ip any idea about such behaviour ? Tnx ! P.S. Fw version is snom190-SIP 3.46 -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 IP 500 and DTMF
Hi all ! I played around for a few hours with a polycom 500 phone and it seems me that the dtmf mode is not configurable, looks like it only has inband mode. While this is ok with G711 I assume that will result in some troubles using G729, altought I cant test it because I havent got any g729 licence yet. Anyone has tried and is willing to share his impressions ? TNX ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ${CALLERID}
Hi, need a quick help ... it should be easy but ... exten =_9898,1,Answer exten =_9898,2,VoiceMailMain([EMAIL PROTECTED]) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer(Zap/8-1, ) in new stack -- Executing VoiceMailMain(Zap/8-1, @domain) in new stack As you can see there variable CALLERID is empty, why ? I tried also with CALLERIDNUM, same result. Tnx for any help . -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ${CALLERID}
Hello Andreas, Thursday, September 2, 2004, 2:28:33 PM, you wrote: AS [EMAIL PROTECTED] wrote: Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer(Zap/8-1, ) in new stack -- Executing VoiceMailMain(Zap/8-1, @domain) in new stack As you can see there variable CALLERID is empty, why ? AS Sending a question again doesn't mean it wil be answered. Sometimes yes, as you can see :) Seriously, I noticed that timing is important in asking for help, when is morning here most of the members of the list are still sleeping AS Anyway, Wait() is your friend I think, because sometimes caller id AS information is not immediately sent. I'd wait one or two seconds AS before dialing out again. Will try this at once, tnx ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ring tone when busy in trunk scenario
Hi,
I have an asterisk box connected to a PRI LINE, some extensions are
trunked by IAX to another box that's connected via ISDN BRI to a PBX.
That's what's happening
call comes in via PRI to the first box and is sent to the other box
exten = _N.,1,Dial(IAX2/sip:[EMAIL PROTECTED]/*${EXTEN})
the other box rings a pbx (simulating an ISDN call from a BRI line)
exten = _*.,1,Dial(Zap/g1/0553035${EXTEN:1})
The problem is that when a call comes in I get a ringing tone after
the trunking call is received from the second box, prior of dialing
out.
If the called extension is busy a busy condition is reflected to the
first box, but still I get the ringing tone till the timeout where I
hang up.
Is there a way to signal a busy condition without answering the PRI call ?
I would like to have
PRI CALL COMING IN (SILENCE)
TRUNK
CALL TO THE EXTENTION
if extension is ringing
Ring tone to PRI CALL (no answer yet)
PRI call answer on called extension pickup
if extension is busy
Busy tone to PRI CALL (no answer at all)
Hoping for help ... tnx !
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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[Asterisk-Users] CLI variable not set on incoming call
Hi, need a quick help ... it should be easy but ... exten =_9898,1,Answer exten =_9898,2,VoiceMailMain([EMAIL PROTECTED]) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer(Zap/8-1, ) in new stack -- Executing VoiceMailMain(Zap/8-1, @domain) in new stack As you can see there variable CALLERID is empty, why ? I tried also with CALLERIDNUM, same result. Tnx for any help . -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Sip Channel CLI
Hello Jason, Friday, August 27, 2004, 12:18:23 PM, you wrote: JW On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi JW [EMAIL PROTECTED] wrote: Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. JW In the general section of sip.conf use the following line JW fromdomain=sip.address.com Tnx ! Do I also have to define a peer in sip.conf or the registration as S exten is sufficient? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Channel CLI
Hello asterisk-users, I have had asterisk registering as a sip extension to an external provider, calls are coming in in pretty fine. Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. I googled around but I have find nothing usefoul by now ... any guess? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting an USA phone number
Hi, it may be off topic, in case excuse me. I need to get a USA phone number, possibly a Florida one. I would like to use my actual sip phone to connect, also I would love montly flat rates on calls. Have you got a provider to suggest that will allow me to subscribe with a credit card (easy) and with my Grandstream ATA sip device (less easy it seems)? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] HFC-S card and Unable to create channel of type 'Zap'
Ciao ! are you connecting a phone or a pbcx to the isdn card ? Hello Maurizio, Tuesday, July 13, 2004, 2:24:24 PM, you wrote: MM -BEGIN PGP SIGNED MESSAGE- MM Hash: SHA1 MM hi, MM i'm new to * MM I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; MM when i try to call outside i get: MM-- Accepting AUTHENTICATED call from 192.168.1.110, MM requested format = 1024, actual format = 1024 MM -- Executing Dial([EMAIL PROTECTED]/2, Zap/g1/0123456) in new stack MM Jul 13 13:42:49 NOTICE[884752]: app_dial.c:559 dial_exec: MM Unable to create channel of type 'Zap' MM == Everyone is busy at this time MM Jul 13 13:43:07 WARNING[163851]: chan_zap.c:6070 MM zt_pri_error: PRI: Read on 19 failed: Unknown error 500 MM Jul 13 13:43:07 NOTICE[163851]: chan_zap.c:6976 pri_dchannel: PRI got event: 6 on span 1 MM - MM /etc/zaptel.conf MM loadzone=it MM defaultzone=it MM span=1,1,3,ccs,ami MM bchan=1-2 MM dchan=3 MM - MM ztcfg -v MM Zaptel Configuration MM == MM SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) MM 3 channels configured. MM - MM /etc/asterisk/zapata.conf MM [channels] MM ; MM ; Default language MM ; MM ;language=en MM ; MM ; Default context MM ; MM ; MM switchtype = euroisdn MM ; p2mp TE mode MM signalling = bri_cpe_ptmp MM pridialplan = local MM prilocaldialplan = local MM echocancel=yes MM immediate=yes MM group = 1 MM context = local channel = 1-2 *CLI zap show channel 1 MM Channel: 1 MM File Descriptor: 17 MM Span: 1 MM Extension: MM Context: local MM Caller ID string: MM Destroy: 0 MM Signalling Type: PRI Signalling MM Owner: None MM Real: None MM Callwait: None MM Threeway: None MM Confno: -1 MM Propagated Conference: -1 MM Real in conference: 0 MM DSP: no MM Relax DTMF: no MM Dialing/CallwaitCAS: 0/0 MM Default law: alaw MM Fax Handled: no MM Pulse phone: no MM Echo Cancellation: 128 taps unless TDM bridged, currently OFF MM PRI Flags: MM Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7351 MM zap_show_channel: Failed to get conference info on channel 1 MM Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7357 MM zap_show_channel: Failed to get confmute info on channel 1 MM any help will be very apreciated MM 10x MM Maurizio -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] zaphfc - hfc pci based ISDN card : point2point DDI
Hello Tomaz, Wednesday, June 30, 2004, 10:58:56 AM, you wrote: T hello, T anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? T what kernel ? Dunno what DDI is but I'm currently using a HFC card in NT mode point2point using the package bristuff 0.0.0.2 with fedora core 1 and his out of the box kernel. T and second question mISDN driver .. anyone has working solution with T mISDN and maybe fritz card? T what you suggest for DDI - point2point mode (card,kernel,chan_..., ...) ? T thank you, T Tomaz T ___ T Asterisk-Users mailing list T [EMAIL PROTECTED] T http://lists.digium.com/mailman/listinfo/asterisk-users T To UNSUBSCRIBE or update options visit: Thttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Hello Robinson, Wednesday, June 30, 2004, 11:19:35 AM, you wrote: RTW We are using the HFC card in point-to-point mode with DDI. RTW I am using bri-stuff-0.0.2 as well. Have someone got a list of bristuff compatible ISDN card ? I have, for example, some DIGI (datafire) cards that have an HFC-S chipset but that are not recnognized by zaphfc some other brands, same chipset but different revision (last number on chip, I guess is revision code) are working ok. Looks strange, no ? :) RTW Rgds RTW Tim RTW -Original Message- RTW From: [EMAIL PROTECTED] RTW [mailto:[EMAIL PROTECTED] On Behalf Of Alessio RTW Focardi RTW Sent: 30 June 2004 10:12 RTW To: Tomaz RTW Subject: Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : RTW point2point DDI RTW Hello Tomaz, RTW Wednesday, June 30, 2004, 10:58:56 AM, you wrote: T hello, T anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? T what kernel ? RTW Dunno what DDI is but I'm currently using a HFC card in NT mode RTW point2point using the package bristuff 0.0.0.2 with fedora core 1 and RTW his out of the box kernel. T and second question mISDN driver .. anyone has working solution with T mISDN and maybe fritz card? T what you suggest for DDI - point2point mode (card,kernel,chan_..., T ...) ? T thank you, T Tomaz T ___ T Asterisk-Users mailing list T [EMAIL PROTECTED] T http://lists.digium.com/mailman/listinfo/asterisk-users T To UNSUBSCRIBE or update options visit: Thttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Hello Robinson, Wednesday, June 30, 2004, 11:35:33 AM, you wrote: RTW There are several cards that use the chipset, and we had to modify the RTW code to get it to recognise the cards. I have spoken to kapejod and the RTW next release 0.0.3 of the drivers (due out this week!!) will support RTW detection of the chipset itself rather than the card ID, so you should RTW find that the new version will fix this. RTW If you want I will send you the patch we made to pick up Asustec cards RTW as well as Billion cards. Tnx Tim, I will wait for version 0.0.3 if its just a matter of days ... I would like also to point out (expecially for future users googling) that I solved a serious voice sync problem on some server adding apm=off acpi=off pci=noacpi apic as kernel boot parameters Regards ! RTW Rgds RTW Tim RTW -Original Message- RTW From: [EMAIL PROTECTED] RTW [mailto:[EMAIL PROTECTED] On Behalf Of Alessio RTW Focardi RTW Sent: 30 June 2004 10:28 RTW To: Robinson Tim-W10277 RTW Subject: Re[2]: [Asterisk-Users] zaphfc - hfc pci based ISDN card : RTW point2point DDI RTW Hello Robinson, RTW Wednesday, June 30, 2004, 11:19:35 AM, you wrote: RTW We are using the HFC card in point-to-point mode with DDI. RTW I am using bri-stuff-0.0.2 as well. RTW Have someone got a list of bristuff compatible ISDN card ? RTW I have, for example, some DIGI (datafire) cards that have an HFC-S RTW chipset but that are not recnognized by zaphfc some other brands, RTW same chipset but different revision (last number on chip, I guess is RTW revision code) are working ok. RTW Looks strange, no ? :) RTW Rgds RTW Tim RTW -Original Message- RTW From: [EMAIL PROTECTED] RTW [mailto:[EMAIL PROTECTED] On Behalf Of Alessio RTW Focardi RTW Sent: 30 June 2004 10:12 RTW To: Tomaz RTW Subject: Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : RTW point2point DDI RTW Hello Tomaz, RTW Wednesday, June 30, 2004, 10:58:56 AM, you wrote: T hello, T anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? T what kernel ? RTW Dunno what DDI is but I'm currently using a HFC card in NT mode RTW point2point using the package bristuff 0.0.0.2 with fedora core 1 RTW and his out of the box kernel. T and second question mISDN driver .. anyone has working solution with T mISDN and maybe fritz card? T what you suggest for DDI - point2point mode (card,kernel,chan_..., T ...) ? T thank you, T Tomaz T ___ T Asterisk-Users mailing list T [EMAIL PROTECTED] T http://lists.digium.com/mailman/listinfo/asterisk-users T To UNSUBSCRIBE or update options visit: Thttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax.conf : what is the purpose of trunk ?
Sorry for the stupid question: What's the purpose of defining a peer as trunk in iax.conf ? The question is also valid generally speaking (for other channel types), for instance: why define a Zap group as trunk in extension.conf ? Tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC ISDN card with bristuff from junghanns.net?
Hi, has anyone succesfully installed such scenario ? I'm having problem with Award bios mb pc's... it do works with others, what's your idea ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et?
Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your RTW problem is? I have 2 cards working in one PC, but have had RTW problems with Dell motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et?
Hello Robinson, Thursday, June 17, 2004, 1:19:12 PM, you wrote: RTW Hi Alessio RTW Yes, the problems you report do seem similar to the issues RTW I had. I found on the Dells that the audio prompts were very RTW choppy and played slower than normal. Occasionally there would RTW be 'bursts' oav a second or so of 'good' audio. RTW I also suspected IRQ issues but the Dell Mobos had no way RTW of adjusting them. Best thing is to try and get the card on its RTW own unshared IRQ. If this fails, you either have to try a RTW different pc, or collect 600 euros together and send them to RTW Junghanns.net, and they will send you a quadBRI card that does RTW not have this problem. Well card has his own irq, I will try to tweak bios parameters to see if something gets better. Meanwhile since I orderer 2 dell's yesterday hoping to solve the problem I'm going to bang my head against the wall until they arrive Tnx for now ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.net?
HS a) an IRQ problems, see cat /proc/interrups HS b) a mainboard problem (because usually you've to change the mainboard to HS change the BIOS) HS In case of a), try disabling built-in peripherals of the board, e.g. the HS second serial port, usb host etc. That should make IRQs free. You can HS also try to install the card into a different slot. I'm pretty sure that the card sits on his own IRQ, anyway I'm going to double check that. I'm running fedora core 1 and asterisk was installed using the script I found in the bristuff 0.0.0.2 package. The problem shows in NT or TE mode, the same hard disk installed on a different pc (with another bios) do work. Also I have verified that the hfc card works perfectly using isdn4linux driver. In the motherboard I can tweak PCI LATENCY TIMER: actualy 64 IRQ MODE: actualy APIC also I have tried with hdparm, setting dma mode 3 and other parameters ... still nothing ! Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trunk ?
Hi, I'm pretty new to asterisk so excuse the stupid question: what is the purpose of defining channels as trunks ? I noticed that you can define Zap groups and IAX connections as trunk, but the purpose is not clear to me ... Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Mine strangest asterisk problem ever ....
Hello Brent, Wednesday, June 9, 2004, 7:13:52 PM, you wrote: BF On Wed, 9 Jun 2004, Alessio Focardi wrote: Asterisk with one HFC isdn card, using the zaptel driver bristuff All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. BF Alessio, BF When I was having similar issues the Digium Support folks reccommended BF using hdparm. hdparm sets hard drive parameters (hence hdparm) BF You can try doing different things with it, but I know that I am currently BF set to level 3 rather than 5 as default with RedHat. I checked hdparm googling around, what parameter have you set to 3 instead of 5 ? I'm still banging my head against the wall, the only fix I found to my problem by now is running a script that put some load on the machine to have voice in sync BF - Brent BF ___ BF Asterisk-Users mailing list BF [EMAIL PROTECTED] BF http://lists.digium.com/mailman/listinfo/asterisk-users BF To UNSUBSCRIBE or update options visit: BFhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
