[Asterisk-Users] Zaptel on Suse 9.0
Hi, I have two WCT100P cards installed on a suse 9.0 box. Installation for Zaptel complains of some unresolved dependencies. The zaptel and wct1xxp modules load without any errors. ztcfg give no problems. The problem is when I start asterisk I get the following error and asterisk shuts down - Nov 24 21:28:15 WARNING[18825]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 24 21:28:15 ERROR[18825]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Nov 24 21:28:15 ERROR[18825]: chan_zap.c:9139 setup_zap: Unable to register channel '1-23' Nov 24 21:28:15 WARNING[18825]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Nov 24 21:28:15 WARNING[18825]: loader.c:429 load_modules: Loading module chan_zap.so failed! My /etc/zaptel.conf file reads loadzone=us defaultzone=us span=1,0,0,d4,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,b8zs bchan=25-47 dchan=48 The /etc/asterisk/zapata.conf reads context = default switchtype = national signalling = pri_cpe group = 1 channel = 1-23 context = default switchtype = national signalling = pri_cpe group = 2 channel = 25-47 I don't know if it is some configuration error, or some problem with zaptel and suse9.0. I would really be grateful if someone could help me in this regards. Thanks and regards, - Ashish Shinde ___ 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 Problem!
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck.I know very well this is not kind a problem discussed
in this group but i try my best and all in vein so finally i am here
hoping you ppl helping me out.I discussed this problem in
asterisk's-users group and adding feedback from asterisk-users group my
configs are
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=192.168.10.193
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
Got SIP Responce 482 Loop Detected back from 192.168.10.193
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
*clisip show registry
Host Username
Refresh State
*clisip show users
Username Secret Authen
Def.Context A/C
101 12345678md5,plaintext
sipNo
*clisip show peers
Name/UsernameHost Mask
Port Status
101/101192.168.10.195255.255.255.255
5060Unmonitored
*clisip show channels
PeerUser/ANRCall IDSeq
(Tx/Rx) LagJitterBuffer
0 active SIP channel(s)
Kindly pointout my mistakes/errors and helping me out.
Any Help Is Highly Appreciated.
Thanks in Advance.
Adnan Ahmed.
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Re: [Asterisk-Users] Zaptel on Suse 9.0
Ashish Shinde wrote: Hi, I have two WCT100P cards installed on a suse 9.0 box. Installation for Zaptel complains of some unresolved dependencies. The zaptel and wct1xxp modules load without any errors. ztcfg give no problems. I'm running a TDM400P on SuSE 9.1 stock install without any problems. When I compiled the zap drivers I didn't get the unresolved dependencies issues though, so therein might lie your problem. snip Nov 24 21:28:15 WARNING[18825]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 24 21:28:15 ERROR[18825]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Nov 24 21:28:15 ERROR[18825]: chan_zap.c:9139 setup_zap: Unable to register channel '1-23' Nov 24 21:28:15 WARNING[18825]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Nov 24 21:28:15 WARNING[18825]: loader.c:429 load_modules: Loading module chan_zap.so failed! I'm not sure about the config file since I don't have the same card, but the output above typically happens when the modules don't load up correctly, or there's some other hardware issue at play. Start debugging one step at a time, maybe you should reinstall the drivers taking care of the unresolved dep issues. Flynn ___ 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] SIP Problem!
If you want to Sip REGISTER your phone to asterisk change the
host=192.168.10.193 section of the [101] section to host=dynamic
Currently you are telling asterisk that sip user 101 is on host
192.168.0.193, which is you asterisk box, so when a call goes to 101,
asterisk sends it to itself and then tries to connect the incoming sip call
to 101, hence the loop :)
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] SIP Problem!
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck.I know very well this is not kind a problem discussed
in this group but i try my best and all in vein so finally i am here
hoping you ppl helping me out.I discussed this problem in
asterisk's-users group and adding feedback from asterisk-users group my
configs are
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=192.168.10.193
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
Got SIP Responce 482 Loop Detected back from 192.168.10.193
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
*clisip show registry
Host Username
Refresh State
*clisip show users
Username Secret Authen
Def.Context A/C
101 12345678md5,plaintext
sipNo
*clisip show peers
Name/UsernameHost Mask
Port Status
101/101192.168.10.195255.255.255.255
5060Unmonitored
*clisip show channels
PeerUser/ANRCall IDSeq
(Tx/Rx) LagJitterBuffer
0 active SIP channel(s)
Kindly pointout my mistakes/errors and helping me out.
Any Help Is Highly Appreciated.
Thanks in Advance.
Adnan Ahmed.
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Re: [Asterisk-Users] zaphfc sound problems
Thomas Jagoditsch schrieb: yeah, would be my next try to use 2.4. . no other idea in sight at this time. thx all for your help, got it running with kernel 2.4.27 now :-) had some troubles identifiying which of the two hfc cards was the internal/external but after some hours i found out. seems to me that isdn is not too forgiving if you connect a NT-modded hfc to the PSTN ;-) for some reasons i would really like to get it running with 2.6.x, but thats not the point now. most important i can now play around with * and build a solution for myself and learn about it. 2.6.x will work sooner or later anyway. now i have some troubles setting up *, but thats for another post ... wbr.tja... ___ 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] Cannot open /dev/dsp
On Wed, 24 Nov 2004, Norman Zhang wrote: Cannot open /dev/dsp: file or directory not found You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. You do not need any sound card if you don't want to use the console channel drivers. Just take a look at your /etc/asterisk/modules.conf and be sure not to load them (noload = chan_oss.so, noload = chan_alsa.so). -- Regards, Tobias Jönsson, Lund SE___ 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] Can't hear playtones?
Hello, I would like the dialing party to know what happened to the call, since asterisk doesnt relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. Ive changed the county setting to NL in indications.conf and created this test extension: Exten = s,1, answer Exten = s, 2, playback(test) Exten = s, 3, playtones(busy) But I cant hear a busy tone on my sip phone, the call is answered, I hear the test file playback, but no busy tone. I tried to enter the values directly into playtones, but that didnt work either. Am I missing something? Kind regards, E. Versaevel ___ 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] call forwarding to gsm phones
Hii, I want to forward calls from an asterisk server to a local gsm network. I have read the wiki pages on various forums. But the thing i want is to receive the call(Voip) from an asterisk server then it should be forwarded to a gsm network again to either a gsm/ PSTN from the gsm network itself. Please post a help. Anyhow, to answer your question: Look for BlueVoice GSM gateway. There are other products like that on the market as well and BlueVoice is a more expensive one. But at the top of my head, this is a name that I recall. It is a box with an ethernet connection, an antenna and a SIP stack built in. You put your SIM card in the box and it is able to dial out for you. Rene Kluwen Chimit ___ 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] Sip test
Result: Failed to resolve callee's address harry gaillac wrote: Hi all, Anybody would be able to call my voicemail just for test sip:[EMAIL PROTECTED] regards harry Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ 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 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] Changing Asterisk Voicemail Storage Location
On Thu, 2004-11-25 at 16:22, Java Rockx wrote:
Can anyone tell me how difficult it would be to change the way asterisk
stores/retrieves user
messages as follows?
Currently mailboxes are in
/var/spool/asterisk/voicemail/{context}
But I need to store messages in a hash to limit the number of directories per
context. All mailbox
extensions are the user's 10-digit phone number (aka, DID). The parts of a
DID are as follows
So my hashing would look like this
/var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line}
And in the {line} directory we would have the usual Asterisk
files/directories for inbox, etc.
We're looking at a large number of mailboxes and this would give us a maximum
of 1 mailboxes
per directory - which plays nice with the Linux file system.
You might look at alternative filesystem formats. Linux file system is
not any file system I've heard of. Most likely you are referring to the
filesystem that you get by default when you do an install and just click
next without understanding the option each step of the way.
Specifically, look at reiserfs, it is very good at handling directories
with large number of files, as frequantly seen in mail servers using
maildir format etc...
I'm not sure I understand all the details, but reiserfs should be
equivalent in speed to a DB at least, I've frequantly seen it
referred to in that way back when I used to subscribe to their mailing
list.
I suppose you might ask the question, is it faster to parse the mailbox
name in userspace and then look up the correct file, or let the kernel
parse the name, and find the file for you
Hope this helps you...
Regards,
Adam
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Re: [Asterisk-Users] Codec control
But yhis doesn't deal with the canreinvite i have 3 call directions 1. Local - Internet / Internet Local - g729 , media through * 2. Local - Local - g711, media not through * 3. Internet - Internet - g729, media not through * Your solution is working ok , except in the 3-td situation. Can this be done som how with peers ? Eric Wieling wrote: Damian Minkov wrote: How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when Phone 1 calls Phone 3 to use g729 ? Because of the problem with disallow= in sip.conf peer sections this may not work the way you expect. This is what I do. [general] diallow=all allow=ulaw allow=g729 [phone1] disallow=all allow=ulaw [phone2] disallow=all allow=ulaw [phone3] disallow=all allow=g729 Now for the trick. Make the PHONE only support the codec you want. i.e. diallow all the codecs on phone1 and phone 2 except for ulaw. On phone 3 disallow all the codecs except for g729. Because of the problems with disallow= in the [happypeer] parts of sip.conf this won't work unless the codecs are specified on the phone. Do NOT allow both ulaw and alaw. I've seen problems with this reported on #asterisk --Eric -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ 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] oh323 compile issue
Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: openh323 dir: make clean apply the oh323 patch configure make opt asterisk-oh323-0.7 dir: make [...] wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel (H323Connection , int, unsigned int, H323AudioCodec )': wrapendpoint.cxx:915: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' wrapendpoint.cxx:916: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] asterisk-oh323-0.7.0]# Someone know what's the problem? Regards -- Daniel ___ 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] Asterisk on a Linksys WRT54G(S)
A noble feat to attempt, but I have to ask, why? How on earth would this be a benefit of any real use other than you happen to own one and say you've done it? On Wed, 24 Nov 2004 00:52:29 +0100, Bastian Schern [EMAIL PROTECTED] wrote: Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian ___ 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 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] Haven't got a clue ...
On Wednesday 24 November 2004 23:45, Asterisk wrote:
Thanks for the tips - however, all I want to do is to play the sound on
the failed call back to the caller :
If the called number is busy, I want them to hear the busy tone
if the called number is out of order, I want them to hear the 3 tone
etc etc
How do I do this without having to check for each error and play an
appropriate file ?
Asterisk 1.0.2 seems to set ${HANGUPCAUSE} to whatever the ISDN hangup cause
was, so you can just do:
[dial-isdn-a]
exten = _0.,1,AgentMonitorOutgoing(n)
exten = _0.,2,CallingPres(0)
exten = _0.,3,SetCIDName(${CALLERIDNAME})
exten = _0.,4,SetCIDNum(555123) ; number changed to protect the guilty!
exten = _0.,5,Dial(Zap/g3/${EXTEN},40)
exten = _0.,6,SetVar(PRI_CAUSE = ${HANGUPCAUSE})
exten = _0.,7,Hangup()
exten = _118.,1,Dial(Zap/g3/${EXTEN},40)
exten = _118.,2,SetVar(PRI_CAUSE = ${HANGUPCAUSE})
exten = _118.,3,Hangup()
I'm fairly new to *, so there may be huge problems with that which I'm not
aware of.
Dan
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[Asterisk-Users] Re:SIP Problem
I am very thankful to you people for helping me as much i imagine but i still
need your help, problem is that i am not be able to dial from my analog phone
conected to fxs card to my sip phone i change my configs but still no result.
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=dynamic
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqnp 102 (Request)
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqnp 102 (Request)
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
Thanks in Advance.
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Re: [Asterisk-Users] Haven't got a clue ...
Asterisk wrote: Thanks for the tips - however, all I want to do is to play the sound on the failed call back to the caller : If the called number is busy, I want them to hear the busy tone if the called number is out of order, I want them to hear the 3 tone etc etc How do I do this without having to check for each error and play an appropriate file ? On a PRI you have to check for each error (or at least each common error) and play a message. ___ 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] Changing Asterisk Voicemail Storage Location
If a patch is developed that will acomplish this division, I am
interested in it.
My company is planning on deplying a massive * network with a central
server providing VM.
This would make the VM server easyer to admin.
--
Christopher Dobbs
Adam Goryachev wrote:
On Thu, 2004-11-25 at 16:22, Java Rockx wrote:
Can anyone tell me how difficult it would be to change the way asterisk stores/retrieves user
messages as follows?
Currently mailboxes are in
/var/spool/asterisk/voicemail/{context}
But I need to store messages in a hash to limit the number of directories per context. All mailbox
extensions are the user's 10-digit phone number (aka, DID). The parts of a DID are as follows
So my hashing would look like this
/var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line}
And in the {line} directory we would have the usual Asterisk files/directories for inbox, etc.
We're looking at a large number of mailboxes and this would give us a maximum of 1 mailboxes
per directory - which plays nice with the Linux file system.
You might look at alternative filesystem formats. "Linux file system" is
not any file system I've heard of. Most likely you are referring to the
filesystem that you get by default when you do an install and just click
next without understanding the option each step of the way.
Specifically, look at reiserfs, it is very good at handling directories
with large number of files, as frequantly seen in mail servers using
maildir format etc...
I'm not sure I understand all the details, but reiserfs should be
equivalent in speed to a DB at least, I've frequantly seen it
referred to in that way back when I used to subscribe to their mailing
list.
I suppose you might ask the question, is it faster to parse the mailbox
name in userspace and then look up the correct file, or let the kernel
parse the name, and find the file for you
Hope this helps you...
Regards,
Adam
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Re: [Asterisk-Users] Codec control
Eric Wieling wrote:
Damian Minkov wrote:
How can i control the codec for the calls. For example I have 3 SIP
phones registered to asterisk
The firs two are in the local area network (behind nat)- I want to
use g711 between them and to connect directly (canreinvite=yes)
and the third is in internet - want all calls to it and from it to
use g729 and media to go through asterisk.
So if Phone 1 calls Phone 2 the codec to be g711, but when Phone 1
calls Phone 3 to use g729 ?
Because of the problem with disallow= in sip.conf peer sections this
may not work the way you expect. This is what I do.
[general]
diallow=all
allow=ulaw
allow=g729
[phone1]
disallow=all
allow=ulaw
[phone2]
disallow=all
allow=ulaw
[phone3]
disallow=all
allow=g729
Now for the trick. Make the PHONE only support the codec you want.
i.e. diallow all the codecs on phone1 and phone 2 except for ulaw. On
phone 3 disallow all the codecs except for g729. Because of the
problems with disallow= in the [happypeer] parts of sip.conf this
won't work unless the codecs are specified on the phone. Do NOT allow
both ulaw and alaw. I've seen problems with this reported on #asterisk
--Eric
Damian Minkov wrote:
But yhis doesn't deal with the canreinvite
i have 3 call directions
1. Local - Internet / Internet Local - g729 , media through *
2. Local - Local - g711, media not through *
3. Internet - Internet - g729, media not through *
Your solution is working ok , except in the 3-td situation. Can this be
done som how with peers ?
You can try playing around with ${SIP_CODEC). Check README.variables to
be sure of the correct variable name. Picking the codec based on the
destination is not well supported in Asterisk. Asterisk assumes you
will want to pick the codec based on the source device of the call, not
the destintion device.
If you want to have the media NOT go thru Asterisk you will have
problems with NAT. Also some Dial options (t and T come to mind,
but there may be others) will prevent Asterisk from releasing the media
stream so the clients can talk directly to each other.
Your needs are complex enough that you might want to investigate using
SER as a SIP proxy. I have designed my (small) VoIP network to not
need SER, but I do not need to pick the codec based on the destination
device and I don't have the devices communicate directly.
--Eric
--
I am seeking part or full time employment in Toronto, The Netherlands,
or Belgium. My preference is part time employment in Toronto with
some telecommuting. Currently located in New Orleans, Louisiana and am
happy to relocate. Contact eric at fnords.org.
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Re: [Asterisk-Users] asterisk 1.0.1
amna saleem wrote: Hi I want to download asterisk v1.0.1,can anybody tell me where can i find this version with zaptel ,zapata ,libpri etc thanx in advance Do you want 1.0.1 or do you want the latest stable release of Asterisk? You can get the following 1.0.2 packages: ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.2.tar.gz ftp://ftp.digium.com/pub/zaptel/zaptel-1.0.2.tar.gz ftp://ftp.digium.com/pub/libpri/libpri-1.0.2.tar.gz --Eric -- I am seeking part or full time employment in Toronto, The Netherlands, or Belgium. My preference is part time employment in Toronto with some telecommuting. Currently located in New Orleans, Louisiana and am happy to relocate. Contact eric at fnords.org. ___ 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] Asterisk on a Linksys WRT54G(S)
Well for example : use SIP on your LAN an use IAX to connect the outside world ... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Bryan Mannos Verzonden: do 25/11/2004 10:04 Aan: Asterisk Users Mailing List - Non-Commercial Discussion CC: Onderwerp: Re: [Asterisk-Users] Asterisk on a Linksys WRT54G(S) A noble feat to attempt, but I have to ask, why? How on earth would this be a benefit of any real use other than you happen to own one and say you've done it? On Wed, 24 Nov 2004 00:52:29 +0100, Bastian Schern [EMAIL PROTECTED] wrote: Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian ___ 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 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 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat___ 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] How to make/recieve call using asterisk when there is a power failure?
Hi, I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. In some situations, even though you may have a ups for the Asterisk, network equipment, channel banks, etc, but your ATA, IP phones which located near to your users and probably not connected to the UPS, so these devices wont be able to function. And even if you have a ups, after an hour or two, your uos will drain out, so how? Though we can have few analog pstn lines as standby, but these lines are mostly use for making outgoing calls rather than receiving incoming calls. For a prolong power failure situation, these lines cant really help much, so businesses will be seriously affected. It is possible to contact the telco to re-direct the incoming calls to the standby analog lines, however, it will generally take couple of hours for the telco to make the switch and very likely there will be a fee involve. I read from this forum that many asterisk implementations had been carried out, I wonder how these implementation take care of the power failure situation? Can someone share the views and implementations? ___ 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] astGUIClient Question
Hi All, can someone give me a short procedure on how to install astGUIClient if there's any. I have Installed asterisk and other required installation. Thanks! Regards John Drayton C. Fule Jr. Systems Engineer Imperium Technologies Inc.(Philippines) ___ 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] How to make/recieve call using asterisk when thereis a power failure?
Title: Message Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange that provides your ISDN-30? My experience from customers has been that none of their current phone solutions worked with power loss so they dont care (not enough to pay the extra). Considering the prolification of mobile phones for emergancy calls during power outages I would agree. my 2p. -Original Message-From: TinKoon [mailto:[EMAIL PROTECTED] Sent: 25 November 2004 09:57To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] How to make/recieve call using asterisk when thereis a power failure? Hi, I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. SNIP This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] How to make/recieve call using asterisk when there is a power failure?
TinKoon wrote: During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. snip There's been quite a bit of discussion on this list about failover scenarios and how to go about handling them; check the list archives for the discussions. You can also look in the Wiki for some examples: http://www.voip-info.org/wiki-Asterisk+failover http://www.voip-info.org/tiki-index.php?page=Failover%20switches http://www.voip-info.org/tiki-index.php?page=Asterisk%20High%20Availability%20Solutions If you're talking about a simple installation with a couple of lines you could use the DPDT Relay solution, look it up in the list archives. For larger installations with mission-critical stuff then you'd want to use failover switches. The Wiki has good resources, look it up there. Flynn ___ 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] Question on IXAy (IAXy actually)
nkb wrote: So, do I still need to have an Asterisk server connected to my IAXy even after I've made provision for it? You can only connect IAXy to an asterisk server. Yours or from a VoIP provider. Like, can I just carry this IAXy around(after provision) and just plug into any broadband connection and start making voip calls via my asterisk provider server? Yes, as long as your service provider or your own server supports IAX2 protocol... Any comments from anyone? ___ 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] internet bandwidth
Hello Michael, Sorry for the late reply. The 17kbps are for the G723.1 at 6.3kbps. The additional overhead which increases the bandwidth usage etc depends on the codec. It's not a fixed overhead in bandwidth for all codecs. You can find a few free codec/bandwidth calculators at: http://www.voipcalculator.com http://www.packetizer.com Best regards, Vlasis. Michael Vogel wrote: Hi! Vlasis Hatzistavrou schrieb: 6.3kbps of G723.1 will become around 17kbps on the IP level without silence suppression because of the additional overhead imposed by protocols like RTP, IP, etc . These 17kbps are they independent from codec? That means a A-LAW with 64kbps has got 64+17=81kbps? BTW: I have seen different descriptions regarding the rate of U-LAW. Is it 64 or 56kbps? If you chose IAX instead of SIP, you will save lots of bandwidth if all (or most) of those 20 calls are directed to the same host. Does IAX save bandwith on single calls as well? Bye! Michael ___ 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 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] ZAP FXS problem - no caller id
Hi, Has anyone seen this problem - Nov 25 18:09:14 WARNING[12923]: chan_zap.c:3463 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I started to get this message after upgrading from 1.0.2 stable to the latest CVS. Hope someone can help me out here. Regards Garry Taylor ___ 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] (for FC3 too) Unable to open master device '/dev/zap/ctl'
On Thu, 2004-11-25 at 04:21 +0100, Patrick wrote: On Wed, 2004-11-24 at 15:22 -0500, Doug Campbell wrote: [snip] I have found that I can overcome this error by just unloading the module and then loading it again: [snip] Or try to wait a few seconds for udev to catch up by putting in between sleep 5. This type of problem is being discussed on the Mandrake Cooker (dev version) list. It also affects other things. The general feeling is that one should use a test loop to check the presence of devices because waiting does not guarantee that it is there. For the moment I've fallen back to the good old way I'll wait 'till udevs is really sorted out, or dropped like devfs :) -- Dave Cotton [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] Connecting a PBX with Asterisk via E1 / PRI
Hello I found in the archives of this list, that the Digium Quad E1 Card is capable of doing the Master for an E1 connection. I could not find any hint if the E100 Card is also capable of doing this. Is this feature the PRI / PRA property or is it the net and cpe property. This is not explained in the datasheets. Anyone information or running setups with this? Jens ___ 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] oh323 compile issue
administrator tootai wrote: Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: Wrong, wrong, wrong! 1) Read the README. 2) Get the right versions of OpenH323/Pwlib. 3) Follow the instructions. Michael. openh323 dir: make clean apply the oh323 patch configure make opt asterisk-oh323-0.7 dir: make [...] wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel (H323Connection , int, unsigned int, H323AudioCodec )': wrapendpoint.cxx:915: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' wrapendpoint.cxx:916: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] asterisk-oh323-0.7.0]# Someone know what's the problem? Regards ___ 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] Question on IXAy (IAXy actually)
Yes, as long as your service provider or your own server supports IAX2 protocol... Any comments from anyone? If you were on the road with an IAXy and it need to be reprovisioned and the server ip changed, SOL unless you can ssh in to the server and do the manual rerpov... ___ 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] How to make/recieve call using asterisk when there is a power failure?
On Thu, 25 Nov 2004, TinKoon wrote: However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. In some situations, even though you may have a ups for the Asterisk, network equipment, channel banks, etc, but your ATA, IP phones which located near to your users and probably not connected to the UPS, so these devices wont be able to function. And even if you have a ups, after an hour or two, your uos will drain out, so how? We use an ups with 12 or 24 hours battery time for the load involved. This is neither advanced nor terribly expensive, just dimension the ups appropriately battery-wise and power-wise. For terminals that require power you can use: * power over Ethernet from the switch * power over Ethernet from mid-line injectors * local ups per station/room If your pstn provider is not a total fly-by-night operation your E1/T1 should operate just fine if the power goes out. You may need to plug the termination equipment at your site into your ups, but they are usually fairly low power devices. If your E1 goes dead during a power outage you should switch provider. Now. Peter ___ 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] No hangup(vpb)
Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus ___ 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] How to make/recieve call using asterisk when thereis a power failure?
On Thu, 25 Nov 2004, Alex Barnes wrote: Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange that provides your ISDN-30? They work just fine if your pstn provider is at all serious. If not, switch. They don't belong in the pstn business anyway. An E1 termination can require local power. In that case you will have to provide backup power to it. Some get their power from the central office, in which case this is not a problem. My experience from customers has been that none of their current phone solutions worked with power loss so they dont care (not enough to pay the extra). Considering the prolification of mobile phones for emergancy calls during power outages I would agree. Man, you must have nice and quiet customers. During a large power outage not too long ago (the first one in many years) our primary network connectivity went down after a while but the E1 isdn phone line was working the whole time. Peter ___ 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] Billing (itemized) in the UK
Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list all of the calls per extension, i.e. from the bill we are able to tell which internal extension made what call to which destination at which date/time, how long this call was in minutes and how much that particular call costs. We would like to reuse the three PSTN lines with the asterisk system, and at present there are no plans to utilize other connectiviy (such as ISDN) - we would like to stick with the three PSTN lines. My understanding is that when the asterisk system is running we won't get any itemized bills any more since the telecom provider has no way of telling from which extension a call originated. Questions: To give the extension information to the telco... How can I configure Asterisk to do send extension information? What signalling do I have to provide for outgoing calls to give extension information the telco? Is there a standard for sending extension numbers (i.e. do I have to send some DTMF digits)? Is there a software / asterisk extension (that works in the UK) that allows asterisk to send extension info? Do I need to buy some equipment that can provide this info to the telco? Which? Where could I find more information on that subject? Thank you very much for your consideration. ___ 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] Forwarding Call
Hello list, I have a question, i want to duplicate some hardware pbx behaviour, and was wondering if it was possible in asterisk to do this, and if so how. I want one phone (device) connected to asterisk to ring whenever a special device is ring (let us say in x100p analog interface) but I want asterisk to only answer the call when the phone is picked up. This is because there are some other phones connected to that line. (I know it sounds strange, and you probably wonder why I need asterisk for this but I need it for some other reasons (there is an ISDN2 line attached to is at well.) So when somebody rings me on the analog line, I want all phones (even the once not connected to asterisk) to ring and be able to pick up the phone but I also want a phone that is connected to asterisk to be able to pick up the phone. Is this possible, and if it is, can somebody send me a example config Thanks in advance, Bart Seresia ___ 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] How to make/recieve call using asterisk when thereis a power failure?
An E1 termination can require local power. In that case you will have to provide backup power to it. Some get their power from the central office, in which case this is not a problem. In our neck of the woods (Australia) the dominant carrier typically deploys one of two solutions: (a) E1/PRI over copper - I haven't seen this for a long while, but I believe they're line powered; (b) E1/PRI over fibre - see these all the time, and the transmission rack they install at the customer site includes batteries and a rectifier which typically provides service for a couple of days in a typical installation. Cheers. ___ 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] oh323 compile issue
Michael Manousos a écrit : administrator tootai wrote: Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: Wrong, wrong, wrong! 1) Read the README. Done 2) Get the right versions of OpenH323/Pwlib. Can't come back to an earlier version 3) Follow the instructions. Give up the test of oh323. You told me to test it, I try it ;-) If my configuration don't meet the required one, too bad. -- Daniel ___ 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] Asterisk Hardware Platform - Intel x86 versus Intel RISC Xscale (ARM)
Hi, I would like to take your advice about which hardware paltform is better for Asterisk - x86 or RISC ? I have the following offers: - Mobile Celeron 733MHz $380 - Xscale 667MHz $330 x86 cost is higher than RISC-solution, but the performance is better. The technical specification for both CPU is the same: External FLASH:1 x MMC/SD VGA/LCD: 1 x VGA/LCD Audio: AC 97 USB: 6 x USB 2.0 Integrated FLASH: 128 MB Integrated RAM:256 MB External RAM: DIMM up to 1 GB Ethernet: 2 x 10/100 SATA: 1 x Serial ATA Mini PCI: 1 x Mini PCI PCI: 1 x PCI for PCI Raiser Card with 3 x PCI Slots Power Supply: Single 48 VDC Power Input (36V - 72V), Power over Ethernet specification (IEEE 802.3af). Jumper selectable to choose the source: power connector (jack) or WAN port. Best Regards, Miroslav Nachev ___ 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] How to make/recieve call using asterisk whenthereis a power failure?
-Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: 25 November 2004 10:54 They work just fine if your pstn provider is at all serious. If not, switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during power outage I was simply speculating on the reasons for what I was told, I suspected that this was old wives tails. An E1 termination can require local power. In that case you will have to provide backup power to it. Some get their power from the central office, in which case this is not a problem. This is what I assumed but since couldn't base this on experience I bowed to louder mouths Man, you must have nice and quiet customers. I think the reason they are quiet is because historically they haven't had phones without power. Further to this I bet they were simply told by their old school PBX provider that such things were black magic and not be used. :-) Thanks for clearing this up and confirming what I believed. I think I will look into adding a UPS and connecting the * and the ISDN wall box and maybe a handful of the DECT phones and the reception phones. This will be a nice mix between cost to functionality I think. Cheers Alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] asterisk and pstn
Thank you very much for the reply. That has made things a good bit clearer. Am I correct in my current understanding: So basically if I want to support approx 100 calls, I would have to purchase a digium PRI card and then pay eircom (or whoever my service provider is) approx 3000 a year for the PRI ISDN connection?? My other option is to have a BRI connection which could support approx 8 calls (calls perline, 4 line card) and pay my service provider alot less for a bri connection. I could also use an fxo card in * connected to my hpine line but that would only support one call at a time. Are these the only main implementation options? Has anyone come across the Skype Voip gateway (VTA1000) or where does that fit in the scheme of things?.are there new options emerging?? Sorry if these questions seem a bit dense but I am doing this as part of a research project and since I am a student who has has no experience working in industry I am clueless when it comes to how such a service would be implemented in practice and the cost of it. I am appreciative of any knowledge passed on from others who have practical know how. Thanks again, Aisling. Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk and pstn Date: Wed, 24 Nov 2004 21:11:02 - In simple terms ISDN is a digital interface to the PSTN as opposed to the analogue RJ11 phone connectors your used to at home (POTS - Plain Old Telephony System). ISDN lines (Integrated Services Digital Network) typically comes in two configurations BRI (Basic Rate has 2B+D channels ie. 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 speech + 1 data). Because ISDN is digital the interface it's more advanced and supports a much wider set of functions and services than POTS. In Ireland Eircom market ISDN as 'hi-speed', you may be familiar with this (this is not ADSL, your not always online). The main suppliers of this type of service would be Eircom, Esat and COLT. Line rental for PRI will be in the order of 3K/month Sorry that should read 3K/annum plus a setup fee (yes - that puts the cost of the card in perspective). To conect your * box to PSTN with BRI/PRI interface you'll need one of digium's cards or an equivalent CAPI card. Br /Kev/ - Original Message - From: Ashling O'Driscoll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 24, 2004 4:17 PM Subject: [Asterisk-Users] asterisk and pstn Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about telephony I am now quite confused. This is what i understand so far. If this is incorrect or if anybody has any ideas as to I could implememnt this in a better/more scalable fashion, I would really appreciate it. I could put a fxo card in my asterisk server and connect this to a telephone line. This would enable sip to pstn calls but only one call at a time. To connect analog phones from the inside network(i.e. the asterisk network) going out I would need an fxs card. Now the problem with the above scenario is that only one call would be allowed at a time. I know I could get an fxo card with a few ports but that would still only allow a few calls. To implement a network where several calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ 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 mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] How to make/recieve call using asterisk when thereis a power failure?
On Thu, Nov 25, 2004 at 09:59:58PM +1100, Bob Purdon arranged a set of bits into the following: An E1 termination can require local power. In that case you will have to provide backup power to it. Some get their power from the central office, in which case this is not a problem. In our neck of the woods (Australia) the dominant carrier typically deploys one of two solutions: (a) E1/PRI over copper - I haven't seen this for a long while, but I believe they're line powered; The one we got installed today (yes, really) at work wasn't line powered, another bloody wall-wart to go with the others. However that was shipped as an OnRamp 30 (ie a euro PRI), whereas the E1 data link is also locally powered, but with power coming from the curb. We're just outside the Melbourne CBD (next to melb-uni if people care) and now have 2 E1's, 1 data, 1 ISDN voice, and about another 1/2 dozen analog trunks. (Plus the two ADSL links) (b) E1/PRI over fibre - see these all the time, and the transmission rack they install at the customer site includes batteries and a rectifier which typically provides service for a couple of days in a typical installation. pgpmEaWewChly.pgp Description: PGP signature ___ 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] Billing (itemized) in the UK
Hi Peter You need to first of all ask your Telco what mechanism it uses with your current switch. The most likely ways are 1) Two stage dialling. 1xxx pause PIN exten dialled number 2) access code 1xxx exten dialled number You need to get the specs for this from Your Communications. It is not clear from the web site... Asterisk will cope perfectly with either solution - you will just need to fiddle a bit with the dial plan. Once we know what you have to send to the telco there are tons of people here who will advise on the Dial command you should use to achieve what you want. Rgds Tim Robinson Ps. Any reason why you chose to stick with the analogue solution? Is this just risk mitigation in the early stages? (this is a valid reason, btw!) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Hoppe Sent: 25 November 2004 10:54 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Billing (itemized) in the UK Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list all of the calls per extension, i.e. from the bill we are able to tell which internal extension made what call to which destination at which date/time, how long this call was in minutes and how much that particular call costs. We would like to reuse the three PSTN lines with the asterisk system, and at present there are no plans to utilize other connectiviy (such as ISDN) - we would like to stick with the three PSTN lines. My understanding is that when the asterisk system is running we won't get any itemized bills any more since the telecom provider has no way of telling from which extension a call originated. Questions: To give the extension information to the telco... How can I configure Asterisk to do send extension information? What signalling do I have to provide for outgoing calls to give extension information the telco? Is there a standard for sending extension numbers (i.e. do I have to send some DTMF digits)? Is there a software / asterisk extension (that works in the UK) that allows asterisk to send extension info? Do I need to buy some equipment that can provide this info to the telco? Which? Where could I find more information on that subject? Thank you very much for your consideration. ___ 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 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] Billing (itemized) in the UK
To give the extension information to the telco... How can I configure Asterisk to do send extension information? [Senad Jordanovic] This greatly depends on your provider... What signalling do I have to provide for outgoing calls to give extension information the telco? [Senad Jordanovic] What PBX are you using currently? Is there a standard for sending extension numbers (i.e. do I have to send some DTMF digits)? [Senad Jordanovic] On POTS lines no. On BRI/PRI yes... Where could I find more information on that subject? [Senad Jordanovic] Try http://www.voip-info.org/tiki-index.php?page=Asterisk Senad Jordanovic Bicom Systems, The complete systems provider www.bicomsystems.com USA 1-212-400-7921 UK 0870 682 782 ___ 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] asterisk and pstn
Hi Ashling - my Eircom account manager rang me last Friday and told me that they've dropped the installation fee for an E1 PRI from the c 3K to zero - nothing. The BRIs are still expensive rental enough - 30.99 / mth / BRI. So 4 BRIs would be 123.96/mth for 8 channels. The PRI with 30 channels would be 264.11 / month. There is also a fractional rate connection which is a PRI with only half the channels available - maybe that offeres better value-for-month over the 4 BRIs but I'm not sure off the top of my head of the pricing for that. (those prices are ex VAT). You would have another option if you have fibre from one of the other carriers (Esat, NTL, etc) - you are in a college so you might have? - they can give you E1 lines over their equipment rather than copper coming from the outside. Derek Ashling O'Driscoll wrote: Thank you very much for the reply. That has made things a good bit clearer. Am I correct in my current understanding: So basically if I want to support approx 100 calls, I would have to purchase a digium PRI card and then pay eircom (or whoever my service provider is) approx 3000 a year for the PRI ISDN connection?? My other option is to have a BRI connection which could support approx 8 calls (calls perline, 4 line card) and pay my service provider alot less for a bri connection. I could also use an fxo card in * connected to my hpine line but that would only support one call at a time. Are these the only main implementation options? Has anyone come across the Skype Voip gateway (VTA1000) or where does that fit in the scheme of things?.are there new options emerging?? Sorry if these questions seem a bit dense but I am doing this as part of a research project and since I am a student who has has no experience working in industry I am clueless when it comes to how such a service would be implemented in practice and the cost of it. I am appreciative of any knowledge passed on from others who have practical know how. Thanks again, Aisling. Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk and pstn Date: Wed, 24 Nov 2004 21:11:02 - In simple terms ISDN is a digital interface to the PSTN as opposed to the analogue RJ11 phone connectors your used to at home (POTS - Plain Old Telephony System). ISDN lines (Integrated Services Digital Network) typically comes in two configurations BRI (Basic Rate has 2B+D channels ie. 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 speech + 1 data). Because ISDN is digital the interface it's more advanced and supports a much wider set of functions and services than POTS. In Ireland Eircom market ISDN as 'hi-speed', you may be familiar with this (this is not ADSL, your not always online). The main suppliers of this type of service would be Eircom, Esat and COLT. Line rental for PRI will be in the order of 3K/month Sorry that should read 3K/annum plus a setup fee (yes - that puts the cost of the card in perspective). To conect your * box to PSTN with BRI/PRI interface you'll need one of digium's cards or an equivalent CAPI card. Br /Kev/ - Original Message - From: Ashling O'Driscoll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 24, 2004 4:17 PM Subject: [Asterisk-Users] asterisk and pstn Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about telephony I am now quite confused. This is what i understand so far. If this is incorrect or if anybody has any ideas as to I could implememnt this in a better/more scalable fashion, I would really appreciate it. I could put a fxo card in my asterisk server and connect this to a telephone line. This would enable sip to pstn calls but only one call at a time. To connect analog phones from the inside network(i.e. the asterisk network) going out I would need an fxs card. Now the problem with the above scenario is that only one call would be allowed at a time. I know I could get an fxo card with a few ports but that would still only allow a few calls. To implement a network where several calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. ---Legal
Re: [Asterisk-Users] oh323 compile issue
administrator tootai wrote: Michael Manousos a crit : administrator tootai wrote: Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: Wrong, wrong, wrong! 1) Read the README. Done 2) Get the right versions of OpenH323/Pwlib. Can't come back to an earlier version Use the OH323STAT flag in the top-level Makefile to build a channel driver with staically linked the libraries if you can't setup multiple OpenH323/Pwlib versions on one machine. 3) Follow the instructions. Give up the test of oh323. You told me to test it, I try it ;-) If my configuration don't meet the required one, too bad. You can do it. Just try harder! Michael. ___ 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] record call on demand
hi all, this is the first time i write to this mailing...I hope to do not wrong and that to speak an understandable English. in these days i have the problem to record calls of a particular extension according to database entry that i can change in every minute. when a call arriving to Asterisk, it should check the db to see if it have to record calls for that extension, if yes it records otherwise * go on. there something that do this or i have to create something for me? someone can send me some info or examples? thanks milk -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.4.2 - Release Date: 24/11/2004 ___ 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] IPv6 and Asterisk?
Hi Jasko: Kphone and Linphone can do v6. We've experienced problems with voice quality using these in the wide area. Not sure this is something to do with playout buffering or other packet handling problem (?). We have also developed a prototype UA based on RAT and the Vovida sipset. It was demonstrated last week at IST 2004 in Den Haag. Regards, -- Socrates UCL - Computer Science +44 20 7679 3696 On Thu, 25 Nov 2004 13:07:20 +0100 (CET), Jasminko Mulahusic [EMAIL PROTECTED] wrote: We can do VoIP tests in the wide area over our native IPv6 connection. Please drop us a line when a v6 branch becomes available in the CVS. SER is IPv6-enabled and we're currently using that as SIP registrar. We then do v4-to-v6 and v6-to-v4 SIP translations to Asterisk using the sip phone numbering scheme. do you know of any v6-capable sip clients? thanx. jasko ___ 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] bristuff'ed version doesn't run
There were some old libs from another installation in /usr/lib/asterisk/modules. rmed them and make installed them again, worked. Alex Mack wrote: Hi everybody! I've managed to compile the bristuff patch on asterisk from Junghanns.net. I want to run this on the quadBRI card built into the PC. As I said, driver and asterisk are compiling well, but asterisk -c bombs with this message: [pbx_dundi.so]Nov 24 15:57:53 WARNING[1076754432]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_sign_bin Nov 24 15:57:53 WARNING[1076754432]: loader.c:423 load_modules: Loading module pbx_dundi.so failed! lcn-asterisk1:~/dl/bri-stuff.0.1.0-RC4a/qozap # Ouch ... error while writing audio data: : Broken pipe Seems to be a problem with an undeclared symbol, but why? Has anybody got - or better: solved - a similar problem? TIA, Alex Mack. ___ 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 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] Call to x-lite clients failing?
Hi, When I call from a SIP phone to X-lite client on a PC (on same local network) I see the following error: -- Executing Dial(SIP/home2-a189, SIP/xlite1|20|tr) in new stack Nov 25 12:45:01 WARNING[1589264]: chan_sip.c:600 __sip_xmit: sip_xmit of 0x973a0bc (len 765) to 192.168.1.27 returned -1: Invalid argument -- Called xlite1 Nov 25 12:45:02 WARNING[114695]: chan_sip.c:600 __sip_xmit: sip_xmit of 0x973a0bc (len 765) to 192.168.1.27 returned -1: Invalid argument Nov 25 12:45:03 WARNING[114695]: chan_sip.c:600 __sip_xmit: sip_xmit of 0x973a0bc (len 765) to 192.168.1.27 returned -1: Invalid argument I can call from the X-lite client to SIP phone ok and I can call between SIP phones ok. Any pointers would be very welcome! thanks Mike ___ 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] Horrible BUZZZZ noise when sounds/music play on SIP phone?
More info on this! I bought a Grandstream budgetphone today. The buzz is not apparent with this phone! If I dial 8500 from the Tecom phone, it tells me its connected to voicemail and then the BUUZZZ, which does not stop! It ignores my DTMF tones for entering the mailbox, just BUUUZZZ. If I do the same from the Grandstream phone it is fine? Weird huh? Thanks Mike On Wed, 24 Nov 2004 23:05:23 +, Mike Dent [EMAIL PROTECTED] wrote: I've just noticed that this sound is not apparent when dialing from X-Lite but is there from the Tecom SIP phone? Does that make any more sense? thanks Mike On Wed, 24 Nov 2004 22:48:36 +0100 (CET), Peter Svensson [EMAIL PROTECTED] wrote: On Wed, 24 Nov 2004, Andrei (MPI) wrote: David Boyd wrote: On Wed, 2004-11-24 at 04:14, Mike Dent wrote: Hi, I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and 1 SIP phone. I've noticed some horrible buzz/rasping type of sounds! These seem to occur when * is trying to play back some audio or sound to me? E.g. If I have an exten rule which plays one of the music on hold files when I dial 800 lets say, I get a really loud buzz for about 2 seconds and then the music plays. E.g. 2. If I dial 500 to connect to Digium, as the call is connecting I get the same loud buzz noise for 0.5 seconds or so. Not sure where this is coming from? I did a search on the wiki for buzz/hum/rasp but could not find anything. Sound like an IRQ issue. Check to see if you are sharing an interrupt on your X100P card, take a peek with cat /proc/interrupts (on linux at least) :) ..and stop running X (x-windows) on your Asterisk box. :) It could be the echo canceler. If the line is completely quiet the echo canceler seems to diverge. I think there was a bug in the bug tracker about this that got fixed a while back. http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002820 The fix was merged on 2004-11-17. It may be a good idea to update and check again. Peter ___ 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 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] Billing (itemized) in the UK
Thank you very much for the answer! I think it is a good path to look at. I have had a look through
our paperwork for the present pbx, and I found one document that seemed to indicate we have to dial
1666extndialled_number
to give the extn info to the telco. The paper is a bit old (1999) and since then we have changed our
telco, but I guess that this protocol is still valid. This afternoon I will hook up a recording
device on the line and see which digits are actually dialled when I dial an outside line. From the
recording I should be able to reconstruct which digits have actually been dialled by the pbx.
If the protocol is correct, I could construct a dial command such as
exten = _9.,1,Dial(Zap/g1/1666ID${EXTEN:1})
or so - I would just need a way to construct id - and then any caller from an inside device would
just prepend a '9' before the real number. I probably would also bar simple '9' dialling to get an
outside line... lets see.
Keep you posted, and so many thanks for all the help!
P
Hi Peter
You need to first of all ask your Telco what mechanism it uses with your
current switch. The most likely ways are
1) Two stage dialling. 1xxx pause PIN exten dialled number
2) access code 1xxx exten dialled number
You need to get the specs for this from Your Communications. It is not
clear from the web site...
Asterisk will cope perfectly with either solution - you will just need
to fiddle a bit with the dial plan. Once we know what you have to send
to the telco there are tons of people here who will advise on the Dial
command you should use to achieve what you want.
Rgds
Tim Robinson
Ps. Any reason why you chose to stick with the analogue solution? Is
this just risk mitigation in the early stages? (this is a valid reason,
btw!)
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 10:54
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Billing (itemized) in the UK
Hello!
We are located in the UK, and we are planning to replace our old pbx
with an asterisk based pbx. For
outgoing calls our present pbx is connected to three PSTN lines which
all have the same number.
Internally, the pbx caters for quite a few extensions, and each
extension can make outbound phone calls.
Our telecom provider (your communications) gives us monthly itemized
bills that list all of the
calls per extension, i.e. from the bill we are able to tell which
internal extension made what call
to which destination at which date/time, how long this call was in
minutes and how much that
particular call costs.
We would like to reuse the three PSTN lines with the asterisk system,
and at present there are no
plans to utilize other connectiviy (such as ISDN) - we would like to
stick with the three PSTN lines.
My understanding is that when the asterisk system is running we won't
get any itemized bills any
more since the telecom provider has no way of telling from which
extension a call originated.
Questions:
To give the extension information to the telco...
How can I configure Asterisk to do send extension information?
What signalling do I have to provide for outgoing calls to give
extension information the telco?
Is there a standard for sending extension numbers (i.e. do I have to
send some DTMF digits)?
Is there a software / asterisk extension (that works in the UK) that
allows asterisk to send
extension info?
Do I need to buy some equipment that can provide this info to the telco?
Which?
Where could I find more information on that subject?
Thank you very much for your consideration.
___
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Re: [Asterisk-Users] asterisk and verizon DSL
On Wednesday 24 November 2004 08:27 pm, Scott Laird wrote: On Nov 24, 2004, at 4:18 PM, [EMAIL PROTECTED] wrote: Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some troubleshooting, but I'm really interested in knowing of anyone's setup that already has Asterisk working with Verizon residential DSL. Mine works okay, but I'm using Verizon's business DSL with a static IP. Scott I have a residential DSL up on a Westel modem in Verizon land. No problem. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] No hangup(vpb)
Altus Snyman wrote: Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why if i'm not mistaken the OpenLine4 cards do not have hardware hangup detect capability -- you've got to program the logic in the vpb driver. that was how it was with the one I bought about a year and a half or so ago, anyways. i'm not sure if the current crop has that capability built-in. flynn ___ 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] asterisk and pstn
On Thu, 25 Nov 2004, Ashling O'Driscoll wrote: So basically if I want to support approx 100 calls, I would have to purchase a digium PRI card and then pay eircom (or whoever my service provider is) approx 3000 a year for the PRI ISDN connection?? 100 simultaneous calls would require 4 E1 based PRIs. That would give you 120 simultaneous calls. For this volume few other options exist if you go for traditional telephony solutions. My other option is to have a BRI connection which could support approx 8 calls (calls perline, 4 line card) and pay my service provider alot less for a bri connection. Or did you mean 10 simultaneous calls? For that eiter a single pri or several bri:s are possible. The pri route may be simpler. I could also use an fxo card in * connected to my hpine line but that would only support one call at a time. Isdn connection gives you a lot more than just greater density of connections. You get answer and disconnect supervision and overall a better call handling. Are these the only main implementation options? Has anyone come across the Skype Voip gateway (VTA1000) or where does that fit in the scheme of things?.are there new options emerging?? In most countries there are one or several operators providing VoIP connections to the pstn. These could provide you with SIP, H.323 or perhaps even IAX based connections over IP. Peter ___ 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] Cannot open /dev/dsp
On Thursday 25 November 2004 12:08 am, Norman Zhang wrote: Cannot open /dev/dsp: file or directory not found That means you probably don't have a soundcard configured. I don't have one in my test box either, but that doesn't prohibit asterisk from starting up. it just means you can't do certain things from the CLI. You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. Try starting up asterisk in verbose mode, a-la: asterisk -vvvc Are all those v's for real? Asterisk supports 9 levels of verbosity. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] No hangup(vpb)
How and what and where? Sorry I'm a bit new to asterisk and programming Thanks Altus el Flynn wrote: Altus Snyman wrote: Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why if i'm not mistaken the OpenLine4 cards do not have hardware hangup detect capability -- you've got to program the logic in the vpb driver. that was how it was with the one I bought about a year and a half or so ago, anyways. i'm not sure if the current crop has that capability built-in. flynn ___ 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 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] Question on IXAy (IAXy actually)
Yes, as long as your service provider or your own server supports IAX2 protocol... Any comments from anyone? IAXy currently supports IAX or IAX2? The specs say IAX, it didnt mention about IAX2, so, is there a newer version of it or there's a way to upgrade the firmware? Or they dont make a different? ___ 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] Voicetronix OpenSwitch
Hi, Is there anybody using Voicetronix openswitch 6 or 12 with Asterisk ??? Can interchange solutions and experiences? Thanks In advance Prof. Marcelo Kruk -- Prof. Marcelo Kruk - System Manager Webmaster - Voip Consultant Colegio Nacional Jose Pedro Varela - Colonia 1637 CP 11200 Phone: +598 2 4097020 Fax: +598 2 4093219 Data: +598 2 4095977* Montevideo Uruguay South America URL: http://www.reu.edu.uy Internet Society Member 1336660 ICANN Member - 218338 Linux User 18471 ___ 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] Changing Asterisk Voicemail Storage Location
reiserfs, ext2, ext3, etc, etc all blow up eventually, although at differnet
capacities.
Therefore simply changing from ext3 to reiserfs is, IMHO, a total band-aide
since it too has
limitations.
Hashing hundreds-of-thousands of directories seems to be to only real
alternative to keeping the
linux file system from blowing up.
Regards,
Paul
--- Adam Goryachev [EMAIL PROTECTED] wrote:
On Thu, 2004-11-25 at 16:22, Java Rockx wrote:
Can anyone tell me how difficult it would be to change the way asterisk
stores/retrieves user
messages as follows?
Currently mailboxes are in
/var/spool/asterisk/voicemail/{context}
But I need to store messages in a hash to limit the number of directories
per context. All
mailbox
extensions are the user's 10-digit phone number (aka, DID). The parts of a
DID are as follows
So my hashing would look like this
/var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line}
And in the {line} directory we would have the usual Asterisk
files/directories for inbox, etc.
We're looking at a large number of mailboxes and this would give us a
maximum of 1
mailboxes
per directory - which plays nice with the Linux file system.
You might look at alternative filesystem formats. Linux file system is
not any file system I've heard of. Most likely you are referring to the
filesystem that you get by default when you do an install and just click
next without understanding the option each step of the way.
Specifically, look at reiserfs, it is very good at handling directories
with large number of files, as frequantly seen in mail servers using
maildir format etc...
I'm not sure I understand all the details, but reiserfs should be
equivalent in speed to a DB at least, I've frequantly seen it
referred to in that way back when I used to subscribe to their mailing
list.
I suppose you might ask the question, is it faster to parse the mailbox
name in userspace and then look up the correct file, or let the kernel
parse the name, and find the file for you
Hope this helps you...
Regards,
Adam
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RE: [Asterisk-Users] Billing (itemized) in the UK
Pete,
I am also in the UK and I have added an include in my extensions.conf for
the file listed bellow.
exten = _15X,1,Dial,${TRUNK}/BYEXTENSION
exten = _147X,1,Dial,${TRUNK}/BYEXTENSION
exten = _NX,1,Dial,${TRUNK}/BYEXTENSION
exten = _01.,1,Dial,${TRUNK}/BYEXTENSION
exten = _07.,1,Dial,${TRUNK}/BYEXTENSION
exten = _08.,1,Dial,${TRUNK}/BYEXTENSION
exten = _09.,1,goto(nogo,1)
You dont need a 9 for a line, you couls also add lines for barred numbers
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peter Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
Thank you very much for the answer! I think it is a good path to look at. I
have had a look through
our paperwork for the present pbx, and I found one document that seemed to
indicate we have to dial
1666extndialled_number
to give the extn info to the telco. The paper is a bit old (1999) and since
then we have changed our
telco, but I guess that this protocol is still valid. This afternoon I will
hook up a recording
device on the line and see which digits are actually dialled when I dial an
outside line. From the
recording I should be able to reconstruct which digits have actually been
dialled by the pbx.
If the protocol is correct, I could construct a dial command such as
exten = _9.,1,Dial(Zap/g1/1666ID${EXTEN:1})
or so - I would just need a way to construct id - and then any caller from
an inside device would
just prepend a '9' before the real number. I probably would also bar simple
'9' dialling to get an
outside line... lets see.
Keep you posted, and so many thanks for all the help!
P
Hi Peter
You need to first of all ask your Telco what mechanism it uses with your
current switch. The most likely ways are
1) Two stage dialling. 1xxx pause PIN exten dialled number
2) access code1xxx exten dialled number
You need to get the specs for this from Your Communications. It is not
clear from the web site...
Asterisk will cope perfectly with either solution - you will just need
to fiddle a bit with the dial plan. Once we know what you have to send
to the telco there are tons of people here who will advise on the Dial
command you should use to achieve what you want.
Rgds
Tim Robinson
Ps. Any reason why you chose to stick with the analogue solution? Is
this just risk mitigation in the early stages? (this is a valid reason,
btw!)
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 10:54
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Billing (itemized) in the UK
Hello!
We are located in the UK, and we are planning to replace our old pbx
with an asterisk based pbx. For
outgoing calls our present pbx is connected to three PSTN lines which
all have the same number.
Internally, the pbx caters for quite a few extensions, and each
extension can make outbound phone calls.
Our telecom provider (your communications) gives us monthly itemized
bills that list all of the
calls per extension, i.e. from the bill we are able to tell which
internal extension made what call
to which destination at which date/time, how long this call was in
minutes and how much that
particular call costs.
We would like to reuse the three PSTN lines with the asterisk system,
and at present there are no
plans to utilize other connectiviy (such as ISDN) - we would like to
stick with the three PSTN lines.
My understanding is that when the asterisk system is running we won't
get any itemized bills any
more since the telecom provider has no way of telling from which
extension a call originated.
Questions:
To give the extension information to the telco...
How can I configure Asterisk to do send extension information?
What signalling do I have to provide for outgoing calls to give
extension information the telco?
Is there a standard for sending extension numbers (i.e. do I have to
send some DTMF digits)?
Is there a software / asterisk extension (that works in the UK) that
allows asterisk to send
extension info?
Do I need to buy some equipment that can provide this info to the telco?
Which?
Where could I find more information on that subject?
Thank you very much for your consideration.
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Re: [Asterisk-Users] Changing Asterisk Voicemail Storage Location
I GREPed the Asterisk 1.0.2 source code last night and only found a few
references to
AST_SPOOL_DIR which indicates that a patch would be rather easy.
So I'll try to do this now and share the patch. I'm thinking of something like
an option in the
Makefile called DID_HASHING, which will enable this, otherwise the current
spool storage system
will be used.
I'll keep you posted.
Regards,
Paul
--- Christopher Dobbs [EMAIL PROTECTED] wrote:
If a patch is developed that will acomplish this division, I am
interested in it.
My company is planning on deplying a massive * network with a central
server providing VM.
This would make the VM server easyer to admin.
--
Christopher Dobbs
Adam Goryachev wrote:
On Thu, 2004-11-25 at 16:22, Java Rockx wrote:
Can anyone tell me how difficult it would be to change the way asterisk
stores/retrieves user
messages as follows?
Currently mailboxes are in
/var/spool/asterisk/voicemail/{context}
But I need to store messages in a hash to limit the number of directories
per context. All
mailbox
extensions are the user's 10-digit phone number (aka, DID). The parts of a
DID are as follows
So my hashing would look like this
/var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line}
And in the {line} directory we would have the usual Asterisk
files/directories for inbox, etc.
We're looking at a large number of mailboxes and this would give us a
maximum of 1
mailboxes
per directory - which plays nice with the Linux file system.
You might look at alternative filesystem formats. Linux file system is
not any file system I've heard of. Most likely you are referring to the
filesystem that you get by default when you do an install and just click
next without understanding the option each step of the way.
Specifically, look at reiserfs, it is very good at handling directories
with large number of files, as frequantly seen in mail servers using
maildir format etc...
I'm not sure I understand all the details, but reiserfs should be
equivalent in speed to a DB at least, I've frequantly seen it
referred to in that way back when I used to subscribe to their mailing
list.
I suppose you might ask the question, is it faster to parse the mailbox
name in userspace and then look up the correct file, or let the kernel
parse the name, and find the file for you
Hope this helps you...
Regards,
Adam
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[Asterisk-Users] oh323/g729 and DTMF
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
I also set inBandDTMF=yes (am not sure if that helped
but nothing happened when i tested again).
Whats the differnce between purchased licences and
passthru mode? I am able to make calls using oh323 and
the codec being used is g729 (since this is the codec
used by our VoIP provider).
But my problem is, the incoming VoIP call seems like
it could not select any keys coz there's no response
(my analysis it is not responding to the DTMF signal).
Anyways, here is part of my extensions.conf under
h323:
[voip-h323]
exten = ${DNIS_TEST},1,Ringing
exten = ${DNIS_TEST},2,Playback(record1)
exten = ${DNIS_TEST},3,Background(silence/3)
exten = 1,1,Goto,nmailbox|s|1
exten = ${DNIS_TEST},4,Dial(Zap/7,5,T)
exten = ${DNIS_TEST},5,Goto,operator1|s|1
exten = ${DNIS_TEST},6,hangup
If you will notice, step 3 will wait for the user to
input 1 if he wants to go to voicemail. This config
works when coming from a PSTN line. But when using
Voip, there is no response.
Lastly, if this is really going nowhere.. Can I use
SIP instead of oh323 in solving this problem of
capturing user's input?? If so, any ideas to go about
it?
If you guys need to view some more of my config, I'd
gladly post it.. =)
Thanks again! and more power!
-Alejandrino
Yahoo! Messenger - Communicate instantly...Ping
your friends today! Download Messenger Now
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[Asterisk-Users] Module Failure
How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. ___ 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] Firefly on Linux
On November 23, 2004 05:28 pm, Adam Hart wrote: iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state --state NEW -j DNAT --to-destination ASTIP iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE Any reason why you need both these statements instead of just a single iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -j DNAT --to-destination ASTIP ?? -A. ___ 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] NEED HELP!!
On November 23, 2004 08:17 am, WipeOut wrote: Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. If your last message had the same kind of super-helpful subject line I would imagine you didn't get any response becuase nobody is going to bother clicking on the message to find out it's something they can or cannot help with. seriously -- in a list with hundreds of messages a day do you really expect that a subject line of NEED HELP!! will get the attention it deserves?? -A. ___ 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] Linksys RT31P2
On November 22, 2004 11:48 am, Kevin P. Fleming wrote: The CPU is only a limitation for a VPN if the pipe the VPN is running over is large/wide. These devices are typically used at the end of a DSL/cable connection, with a maximum bandwidth of a few megabits per second. I don't think that a 200MHz Geode or ARM will have any trouble keeping up with that amount of traffic. It's not the amount of traffic that I'm concerned about, it's having a 200MHz processor processing the encryption for a half dozen or more VPN tunnels -- passing traffic isn't an issue, it's all the processor use for encrypting and decrypting. :-) -A. ___ 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] Billing (itemized) in the UK
You just need to do something like
exten = _9.,1,Dial(Zap/g1/1666$CALLERIDNUM${EXTEN:1})
You can also do some useful translations like
exten = _9[2-8]XX,1,Dial(Zap/g1/1666$CALLERIDNUM0113${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the extension number, followed by the 0113 area code.
You will need to make sure that 999 and 112 go direct to BT by using
another line in the extensions file. E.g.
exten = ,1,Dial(Zap/g1/999)
exten = 9112,1,Dial(Zap/g1/112)
And probably
exten = 999,1,Dial(Zap/g1/999)
Just to be on the safe side!
You could also write a little macro to kick another user off their call
to allow the emergency call to get priority.
There is just so much cool stuff you can do. But do test well!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
If the protocol is correct, I could construct a dial command such as
exten = _9.,1,Dial(Zap/g1/1666ID${EXTEN:1})
or so - I would just need a way to construct id - and then any caller
from an inside device would
just prepend a '9' before the real number. I probably would also bar
simple '9' dialling to get an
outside line... lets see.
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Re: [Asterisk-Users] Module Failure
On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote: How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. Depends on the module. Some modules are very important and can't be skipped. If it is not a module you care about, in the modules.conf, put a noload=module_name into the file and you will get asterisk to skip over it. I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. -- Steven Critchfield [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] asterisk and pstn
If this is to gain knowledge a good source of background information is the IP telephony cookbook http://www.informatik.uni-bremen.de/~prelle/terena/, you should find some answers there. One solution you did not mention is the use of a 3rd party VOIP-PSTN/PLMN gateway - ie. you connect using H.323/SIP/IAX/whatever and they have the PSTN/PLMN interface hardware. Br /Kev/ - Original Message - From: Ashling O'Driscoll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 25, 2004 11:13 AM Subject: Re: [Asterisk-Users] asterisk and pstn Thank you very much for the reply. That has made things a good bit clearer. Am I correct in my current understanding: So basically if I want to support approx 100 calls, I would have to purchase a digium PRI card and then pay eircom (or whoever my service provider is) approx 3000 a year for the PRI ISDN connection?? My other option is to have a BRI connection which could support approx 8 calls (calls perline, 4 line card) and pay my service provider alot less for a bri connection. I could also use an fxo card in * connected to my hpine line but that would only support one call at a time. Are these the only main implementation options? Has anyone come across the Skype Voip gateway (VTA1000) or where does that fit in the scheme of things?.are there new options emerging?? Sorry if these questions seem a bit dense but I am doing this as part of a research project and since I am a student who has has no experience working in industry I am clueless when it comes to how such a service would be implemented in practice and the cost of it. I am appreciative of any knowledge passed on from others who have practical know how. Thanks again, Aisling. Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk and pstn Date: Wed, 24 Nov 2004 21:11:02 - In simple terms ISDN is a digital interface to the PSTN as opposed to the analogue RJ11 phone connectors your used to at home (POTS - Plain Old Telephony System). ISDN lines (Integrated Services Digital Network) typically comes in two configurations BRI (Basic Rate has 2B+D channels ie. 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 speech + 1 data). Because ISDN is digital the interface it's more advanced and supports a much wider set of functions and services than POTS. In Ireland Eircom market ISDN as 'hi-speed', you may be familiar with this (this is not ADSL, your not always online). The main suppliers of this type of service would be Eircom, Esat and COLT. Line rental for PRI will be in the order of ?3K/month Sorry that should read ?3K/annum plus a setup fee (yes - that puts the cost of the card in perspective). To conect your * box to PSTN with BRI/PRI interface you'll need one of digium's cards or an equivalent CAPI card. Br /Kev/ - Original Message - From: Ashling O'Driscoll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 24, 2004 4:17 PM Subject: [Asterisk-Users] asterisk and pstn Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about telephony I am now quite confused. This is what i understand so far. If this is incorrect or if anybody has any ideas as to I could implememnt this in a better/more scalable fashion, I would really appreciate it. I could put a fxo card in my asterisk server and connect this to a telephone line. This would enable sip to pstn calls but only one call at a time. To connect analog phones from the inside network(i.e. the asterisk network) going out I would need an fxs card. Now the problem with the above scenario is that only one call would be allowed at a time. I know I could get an fxo card with a few ports but that would still only allow a few calls. To implement a network where several calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information
Re: [Asterisk-Users] Module Failure
On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote: How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. Depends on the module. Some modules are very important and can't be skipped. If it is not a module you care about, in the modules.conf, put a noload=module_name into the file and you will get asterisk to skip over it. I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. what do you mean with I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. -- Steven Critchfield [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 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] How to make/recieve call using asterisk when thereis a power failure?
We use several Dell 2650 servers. Order them with the dual DC power supply option. Buy a row of -48 batteries and a -48 power source, your servers will stay up for hours. - Original Message - From: TinKoon To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 25, 2004 3:56 AM Subject: [Asterisk-Users] How to make/recieve call using asterisk when thereis a power failure? Hi, I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. In some situations, even though you may have a ups for the Asterisk, network equipment, channel banks, etc, but your ATA, IP phones which located near to your users and probably not connected to the UPS, so these devices wont be able to function. And even if you have a ups, after an hour or two, your uos will drain out, so how? Though we can have few analog pstn lines as standby, but these lines are mostly use for making outgoing calls rather than receiving incoming calls. For a prolong power failure situation, these lines cant really help much, so businesses will be seriously affected. It is possible to contact the telco to re-direct the incoming calls to the standby analog lines, however, it will generally take couple of hours for the telco to make the switch and very likely there will be a fee involve. I read from this forum that many asterisk implementations had been carried out, I wonder how these implementation take care of the power failure situation? Can someone share the views and implementations? ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] oh323/g729 and DTMF
Al Escasa wrote:
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
What I meant was that inband DTMFs do not work when G.729 is used.
Out-of-band DTMFs work just fine. You could send me your config
with a screen log (with -cd options) when you make H.323 calls
to check if there is something weird.
I also set inBandDTMF=yes (am not sure if that helped
but nothing happened when i tested again).
Whats the differnce between purchased licences and
passthru mode? I am able to make calls using oh323 and
the codec being used is g729 (since this is the codec
used by our VoIP provider).
But my problem is, the incoming VoIP call seems like
it could not select any keys coz there's no response
(my analysis it is not responding to the DTMF signal).
Anyways, here is part of my extensions.conf under
h323:
[voip-h323]
exten = ${DNIS_TEST},1,Ringing
exten = ${DNIS_TEST},2,Playback(record1)
exten = ${DNIS_TEST},3,Background(silence/3)
exten = 1,1,Goto,nmailbox|s|1
exten = ${DNIS_TEST},4,Dial(Zap/7,5,T)
exten = ${DNIS_TEST},5,Goto,operator1|s|1
exten = ${DNIS_TEST},6,hangup
If you will notice, step 3 will wait for the user to
input 1 if he wants to go to voicemail. This config
works when coming from a PSTN line. But when using
Voip, there is no response.
Lastly, if this is really going nowhere.. Can I use
SIP instead of oh323 in solving this problem of
capturing user's input?? If so, any ideas to go about
it?
If you guys need to view some more of my config, I'd
gladly post it.. =)
Thanks again! and more power!
-Alejandrino
Michael.
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RE: [Asterisk-Users] Asterisk on a Linksys WRT54G(S)
Well for example : use SIP on your LAN an use IAX to connect the outside world ... Yes, I'll second that need. -Michael ___ 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] Module Failure
On Thu, 2004-11-25 at 12:21 -0300, Francisco Seratti wrote: On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote: How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. Depends on the module. Some modules are very important and can't be skipped. If it is not a module you care about, in the modules.conf, put a noload=module_name into the file and you will get asterisk to skip over it. I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. what do you mean with I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. Exactly what was said. But for those with out the networking background, any open network port has the potential to be exploited. Mark and all do their very best to make sure the code is tight and clean. So if you turn off the VoIP modules not being used, you don't open up more ports to potentially be compromised. -- Steven Critchfield [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] astcc newbie question
I'm trying out ASTCC. I set the card length to 10, and generated a test card. 10 digits. I set the extensions file to: exten = 9175954700,1,Answer exten = 9175954700,2,DeadAGI(astcc.agi) exten = 9175954700,3,Hangup I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand to 9175954700. Was that the right thing to do? Maybe it's not recognizing the DID? Thanks, Bill Hamlin ___ 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] configuring voicemail
i was looking but i dont find how do this: configure the password for the extensions read the messages and some other things related with this can some bady help me with some material or a explicit example. thanks in advance Rodney Acosta Coya. ___ 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] How to make/recieve call using asterisk when thereis a power failure?
I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a huge ups, you will not be able to make and receive any call during power failure, since there will be no power to the Asterisk server. And since all the incoming lines, be it analog lines or T1/E1 are connected to the Asterisk, these lines wont be able to function at all. See google for powerfail transfer switch. Companies like Valcom make them. This is how you handle it on a hardware PBX. It sits in front of the pbx, and on powerfail switches over to an alternate output where you'd hang POTS phones. ___ 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] How to make/recieve call using asterisk when thereis a power failure?
We use several Dell 2650 servers. Order them with the dual DC power supply option. Buy a row of -48 batteries and a -48 power source, your servers will stay up for hours. That's only half of the solution. How will the phones be powered? Some thoughts: -If your power is iffy and youknow it will go out, install a positive interlock at the main breaker panel for those breakers responsible for your servers. Then, get ye to the home reno store and buy a 2kw generator. When the power goes out, the sequence is like this: Power goes out UPS keeps up servers You start the generator Interlock switches from mains to generator UPS "thinks" power is back About $400 for the interlock and, say, $1200 for the generator -Midspan POE injector will keep up your phones if it is UPS'ed as above. Some phones suck a lot of power. My 3Com midspan injector is 200W with 24 ports, but I can only use 15 of those for my Mitel 5220's before the injector shuts down. The 5220's take 14 W at a time! $900 for the injector -A small UPS for each phone would work, they are very inexpensive these days and should keep a phone running for a few hours. $29 at Costco X # of phones -We have a disaster recovery plan with our telco filed. When our T1 goes out, we call a certain number (by cell of course!) and say: "Enable the plan" and they re route our DID's to cell numbers. This is a software change on their end and it takes a few minutes. This will vary by carrier, of course. Ours (Allstream / AT T) are really good to work with. The ILEC, Telus? Not so much. ___ 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] Cannot get two TE410Ps to operate correctly in the same machine
I've been running Asterisk for months with no problems. I have grown to the point where I need an aditional TE cards. After many attempts I was able to add the second card without affecting the performance of the first. However, the second card is not working properly. Setup = - Running on a Dell Server. - RedHat 8 - Asterisk 1.0 - I have two TE410Ps on that box, card 0 assigned to IRQ 4 and card 1 assigned to IRQ 3. - The rotary switch on both cards is set to 0. - I have tried setting the switch on card 1 to other than 0, but it does not work (what is the switch for??) - Both cards have 3 E1 spans and one T1, both in the same order. - Card 0 has worked well for months, and works well even now that card 1 is installed. - zttool finds the 2 cards. Interrupts are as follows: CPU0 0:1382239 XT-PIC timer 1: 4177 XT-PIC keyboard 2: 0 XT-PIC cascade 3:2659119 XT-PIC t4xxp 4:2659098 XT-PIC t4xxp 8: 1 XT-PIC rtc 10: 200713 XT-PIC eth0, eth1 11: 0 XT-PIC ide2 12: 20 XT-PIC PS/2 Mouse 14: 12541 XT-PIC ide0 NMI: 1 ERR: 0 Problems - Choppy voice on calls between channels of card 1. - Even worse on calls between card 0 and card 1. - Card 0 behaves well. - IRQ misses for card 1. Have tried different interrupts. Same thing. - HDLC overrun messages on console for card 1. Strange fact, may be the cause of the problem = Configuration for first span on card 1 is: span=5,1,0,ccs,hdb3 bchan=118-132 dchan=133 bchan=134-148 However, zttool reports card as Internally Clocked. No matter how I've tried, I cannot get card 1 to clock from the external source: Sync Source:Internally clocked First span on card 0 is configured just the same: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 This card gets clocked OK: Sync Source:TE410P (PCI) Card 0 Span 1 Thanks in advance Fernando ___ 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] NEED HELP!!
Andrew Kohlsmith wrote: seriously -- in a list with hundreds of messages a day do you really expect that a subject line of NEED HELP!! will get the attention it deserves?? Sure, it will get _exactly_ the attention it deserves: none. :-) ___ 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] Linksys RT31P2
Andrew Kohlsmith wrote: It's not the amount of traffic that I'm concerned about, it's having a 200MHz processor processing the encryption for a half dozen or more VPN tunnels -- passing traffic isn't an issue, it's all the processor use for encrypting and decrypting. :-) That's exactly what I referring to as well. The number of tunnels does not matter (to any significant degree); all that matters is what size the pipe that the device is connected is. So, if the router is connected to a 1.5Mbps symmetric DSL line, it only has to be able encrypt/decrypt a total of 3Mbps of traffic, period. If a 200MHz ARM or Geode can't do that, I'd be surprised. I know that people are running software-based encryption on 266MHz Geodes and getting far better throughput than that. ___ 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] configuring voicemail
Rodney, thanks for your question. You don't actually have to configure voicemail passwords per extension, but per voicemail box. This is done in the config file 'voicemail.conf'. Here you define each voicemail box number, an associated password and a name (and potentially other parameters). When asterisk starts it reads the voicemail.conf (this activates the settings). Then a user can access his/her voicemail from any device and specify the mailbox and the password from there. Asterisk comes with a voicemail application called 'commedian mail' which offers many options. For more info see http://www.digium.com/asterisk_handbook/voicemail_voicemailmain.html http://www.voip-info.org/tiki-index.php?page=Asterisk+config+voicemail.conf http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain http://www.voip-info.org/wiki-Asterisk+VoiceMail This is just a small selection of links. There are more links on the voip-info pages. I hope this helps! P Rodney Acosta Coya wrote: i was looking but i dont find how do this: configure the password for the extensions read the messages and some other things related with this can some bady help me with some material or a explicit example. thanks in advance Rodney Acosta Coya. ___ 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 -- There are 10 kinds of people in the world, those who understand binary, and those who don't. ___ 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] Module Failure
How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. Depends on the module. Some modules are very important and can't be skipped. If it is not a module you care about, in the modules.conf, put a noload=module_name into the file and you will get asterisk to skip over it. I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. what do you mean with I think it is important to do that for VoIP modules not in use as it also cuts down on potential exploit in routes. Exactly what was said. But for those with out the networking background, any open network port has the potential to be exploited. Mark and all do their very best to make sure the code is tight and clean. So if you turn off the VoIP modules not being used, you don't open up more ports to potentially be compromised. And, not only the port issue but there obviously are a boat load of folks that don't understand default contexts that result in exposures that can be compromised. Would be kind of interesting to do a slow scan of all internet IP's looking for udp 5060 4569 though. :) Rich ___ 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] Cannot get two TE410Ps to operate correctly in the same machine
Problems - Choppy voice on calls between channels of card 1. - Even worse on calls between card 0 and card 1. - Card 0 behaves well. - IRQ misses for card 1. Have tried different interrupts. Same thing. - HDLC overrun messages on console for card 1. Almost sounds like the classic interrupt latency / pci controller issue where there's not enough processor time left to adequately handle the second card. (Obviously that's a pure guess though.) Strange fact, may be the cause of the problem = Configuration for first span on card 1 is: span=5,1,0,ccs,hdb3 bchan=118-132 dchan=133 bchan=134-148 However, zttool reports card as Internally Clocked. No matter how I've tried, I cannot get card 1 to clock from the external source: Sync Source:Internally clocked First span on card 0 is configured just the same: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 This card gets clocked OK: Sync Source:TE410P (PCI) Card 0 Span 1 Realistically, there should only be a single external span interface on your system that is used for sync source. I'm not a programmer so can't verify the code might have limits built in to ensure that. ___ 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] Billing (itemized) in the UK
Thank you very much for the answers - I have hooked up a special adapter and active loudspeaker on
each of the three BT lines, but when I got a line and dial a number I cannot hear any other digits
than those I dial - I would have expected something like seven DTMF bursts/digits (16662xx) before
my digits are audible. Near the pbx I have noticed a small white box saying 'Smiths communications'
and 'SC14' on the lid. The box is connected to two cables - one to a power supply, the other is a 4
pair telephone installation cable with 3 pairs connected. Next to the box is a switch with some
labels on it: one label says 'LINE 1'. The other two labels describe the switch settings - 'SYSTEM'
and 'A/PH MOD'. I have the suspicion that the white box has something to do with the billing and
that it sends some fast data over one of the lines when an outside call is initiated, but I am not
sure. I'll continue to hunt.
I also asked the telecom provider but they were not very helpful and couldn't
(or didn't wish to)
give me any information as to the technical details.
I'll hunt on...
P
Robinson Tim-W10277 wrote:
You just need to do something like
exten = _9.,1,Dial(Zap/g1/1666$CALLERIDNUM${EXTEN:1})
You can also do some useful translations like
exten = _9[2-8]XX,1,Dial(Zap/g1/1666$CALLERIDNUM0113${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the extension number, followed by the 0113 area code.
You will need to make sure that 999 and 112 go direct to BT by using
another line in the extensions file. E.g.
exten = ,1,Dial(Zap/g1/999)
exten = 9112,1,Dial(Zap/g1/112)
And probably
exten = 999,1,Dial(Zap/g1/999)
Just to be on the safe side!
You could also write a little macro to kick another user off their call
to allow the emergency call to get priority.
There is just so much cool stuff you can do. But do test well!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
If the protocol is correct, I could construct a dial command such as
exten = _9.,1,Dial(Zap/g1/1666ID${EXTEN:1})
or so - I would just need a way to construct id - and then any caller
from an inside device would
just prepend a '9' before the real number. I probably would also bar
simple '9' dialling to get an
outside line... lets see.
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There are 10 kinds of people in the world,
those who understand binary, and those who don't.
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Re: [Asterisk-Users] Cannot get two TE410Ps to operate correctly in the same machine
On Thu, 25 Nov 2004, Rich Adamson wrote: However, zttool reports card as Internally Clocked. No matter how I've tried, I cannot get card 1 to clock from the external source: Sync Source:Internally clocked First span on card 0 is configured just the same: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 This card gets clocked OK: Sync Source:TE410P (PCI) Card 0 Span 1 Realistically, there should only be a single external span interface on your system that is used for sync source. I'm not a programmer so can't verify the code might have limits built in to ensure that. According to Marc Spencer on the bug tracker each card is clocked separatly. He recommended the user to contact Digium support. Support for the hardware is included in the price for the Digium cards. Peter ___ 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] Area Code 514 DIDs
Hello, Does anyone here have DIDs for 514 area code - Montreal, QC, in or around the 591 NXX? -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 Blank Bkgrd.gif___ 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] Cannot open /dev/dsp
Cannot open /dev/dsp: file or directory not found You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. You do not need any sound card if you don't want to use the console channel drivers. Just take a look at your /etc/asterisk/modules.conf and be sure not to load them (noload = chan_oss.so, noload = chan_alsa.so). Thanks. After disabling chan_oss.so and chan_alsa.so, the above error message is gone. However I still cannot start asterisk. I'll start another thread. Regards, Norman Zhang ___ 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] Newbie Question
I am just learing some Linux and have been able to setup Asterisk samples and channels fxo card on ch.1 and fxs on ch 4. I have an Internet Polycom phone to use to test to/from internet and 1 analouge phone connected to port 4 of Digium TDM-400 with appropriate cards installed to dial out on. I wish to dial to the outside via PTSN line. I am lost on the instructions. Can anyone help with Extensions.conf and sap.conf. 3 extensions are needed. Thanks for help. Leo ___ 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] Asterisk on a Linksys WRT54G(S)
Michael Devenijn wrote: Well for example : use SIP on your LAN an use IAX to connect the outside world ... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Bryan Mannos Verzonden: do 25/11/2004 10:04 Aan: Asterisk Users Mailing List - Non-Commercial Discussion CC: Onderwerp: Re: [Asterisk-Users] Asterisk on a Linksys WRT54G(S) A noble feat to attempt, but I have to ask, why? How on earth would this be a benefit of any real use other than you happen to own one and say you've done it? On Wed, 24 Nov 2004 00:52:29 +0100, Bastian Schern [EMAIL PROTECTED] wrote: Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian .it's $99 or less. fits in very small space. has highspeed wireless. has a 4 port switch. and has the processing power and ram to handle probably 2-4 calls depending on what you toss at it. now you tell me why people *shouldn't* ___ 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] astGUIClient Question
It is best to go through the SCRATCH_INSTALL that is listed on the project website: http://astguiclient.sf.net/ MATT--- -Original Message- From: john drayton fule [mailto:[EMAIL PROTECTED] Sent: Thursday, November 25, 2004 5:02 AM To: asterisk mailing list Subject: [Asterisk-Users] astGUIClient Question Hi All, can someone give me a short procedure on how to install astGUIClient if there's any. I have Installed asterisk and other required installation. Thanks! Regards John Drayton C. Fule Jr. Systems Engineer Imperium Technologies Inc.(Philippines) ___ 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 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] Opinions on renice or turning off swap or ramdis k as swap?
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? I'm using 2.6 kernel. I've modified the PCI latency in rc.local: setpci -v -s my T100P address latency_timer=ff Anyone else have any performance tips? ___ 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] Opinions on renice or turning off swap or ramdis k as swap?
On Thu, 2004-11-25 at 11:02 -0700, Colin Anderson wrote: I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: 1. asterisk -p == renice -20 ?? Unless you have done something not very smart like putting a DB on your asterisk machine, reniceing asterisk isn't going to give you more clock cycles. 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) What did you expect to get by that? Linux will swap out anything not being used but will keep what it needs. What you have done is make it more likely in a crunch, your machine will fall over instead of trying to gracefully handle the load. 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? Swap in ram? Why not use the ram for ram? I'm using 2.6 kernel. I've modified the PCI latency in rc.local: setpci -v -s my T100P address latency_timer=ff Anyone else have any performance tips? Sounds like you need a beginers book on OS design or even a simple linux/unix admin book. -- Steven Critchfield [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] SIP Phones-Receptionist Setup
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote: Tracy R Reed wrote: On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly: This does seem to be a common request, but I haven't seen any great Yes, it is. I am surprised * still can't do it. I'm not surprised. Asterisk is a PBX, not a key system or a hybrid system. The kind of functionality that is being described here is one or both of those 'other' beasts. Now I'm not saying that this wouldn't be nice, or even a long term requirement if you really want to open the entire SME market, but it's not typical PBX behavior. I would like you to name one PBX that does not support this behavior? Every system from Avaya including their Definity, Merlin Legend, Merlin Magix, Partner, and their new IP based PBXes support it, as do those from Mitel, Nortel, InteCom and every other system that I have ever used. A typical example is a manager/admin setup that works as follows: Sarah a manager has a phone on her desk with call appearances for her main number (x-3123). She also has a phone on her office conference table with its own number (x-3302) but also with shared call appearances for her main number (x-3123). She shares a conference room with Ed, John, Steve, Susan and Simon. All their phone numbers have shared call appearances that conference room's phone. Molly (Sarah's administrative assistant) has a phone with shared call appearances for Sarah, Ed and Susan (two other Executive Team members for whom she provides shared coverage with Wendy and Lisa). When a call comes in for Sarah on x-3123, Molly can answer it, and just by looking at those little red and green lights on her phone she can tell if Sarah is on a call or not. She can then place this call on hold (not park it, just hit that red hold button) and call Sarah announcing this call. Sarah can answer this call just by pressing that button next to the flashing light (indicating a call on hold) and picking up her phone. She does not have to use call pick up. She can also pick this call up on her office conference table, or in the Executive Team's conference room in exactly the same way, not needing to understand or know anything else (press the button with my name on it next to the blinking green led). All of this was done using a PBX (an Avaya Definity), never using call pickup, or an operator console (just a standard 28 button phone for Molly, Wendy and the Executive Team conference room, and a standard 10 button phone for Sarah, Steve, Ed, John, and Simon). This is a real example at a real company, not just something made up as a straw man. If you want to see examples of this, I would be happy to take you to the Math Department at University of Illinois (Nortel), Sony Pictures Imageworks (Avaya) or Argonne National Laboratory's Energy and Environmental Systems group (InteCom). In fact, if you start looking at *all* the differences in functionality, (i.e. call announce, hands free answer-back, hold/pickup scenarios, etc.) it *may* be easier to have a different product stream that is targeting this sort of thing. Of course that's easy to say, but hard to do given the number of developers that are actually working/contributing to * on a regular basis. I would still like to understand how adding any of these features (even if they were not already available on almost every PBX system sold today), would comprise Asterisk's PBXness in some way that would hurt its adoption. This isn't unique to *, it's the battle that every PBX vendor fights at least internally with product management. Yes, but every other PBX vendor has adopted this functionality, while Asterisk has not. How to be all things to all people and still have some level of control over the product development and support streams. I guess what I'm ultimately pointing to is the need to pre-qualify a prospect before one makes a sales proposal. This religious argument (We cannot do that because it is unPBX-like.) seems to also miss another important factor. While large and small organizations use this functionality, a system is almost unusable for a small office without it (see how it is used in every small store or company with a Merlin Legend or Magix system for example). I am fairly convinced that smaller offices are better candidates to adopt Asterisk than are fortune 500 companies. Not having these features makes Asterisk much less likely to be deployed in those environments. While Pingtel's open source sipXchange is not quite ready (still a month or two off from what I have seen), it is getting quite close. I think seeding this whole market segment to them is not the best plan. If there are certain aspects of PBX vs. Key System that they can't metabolize, or aren't willing to make the user training investment, then sell them what they will can rather than try to pound a square peg into their round hole. Does this limit the market for *? Sure does. But hen no
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Nov 20, 2004, at 11:05 PM, Gregory Junker wrote: Most customers don't want to be in a new era. They want something they are accustomed to. I don't need any more impediments to making money than I've already got. So if the customer wants a busy lamp, I am going to do my best to give it to them. I agree. This is why engineers do not make good salesmen. It is also why engineers make poor product marketing managers. While there maybe many more interesting and flashy solutions that offer much more power (We could display current call duration and average call duration over 1 hour, 1 day, 1 week and one month next to each user's name allowing a receptionist to tell a caller how long an average wait time might be.) they are often not what a product's user want or need. On Nov 21, 2004, at 2:49 AM, Peter Svensson wrote: On Sat, 20 Nov 2004, Brian Roy wrote: I would look at putting a dual monitor on her desk. You can pick up a 15 flat panel and a video card for about the same cost as the SNOM. Not to mention, you get quite a bit more benifite from the FOP controls than you do busy lamp fields. It's a a new era here folks. Asterisk is not your dad's pbx. Most people here seem to miss the point that a dedicated hard interface is a lot easier to use than any computer interface. ... You should always design an interface around a human being. A hard interface with a light and a button per extension and so on is really a very good interface. We software pople tent to forget the value of a proper hardware solution. Peter I could not agree more. I think it would be great to have some other options (like an embeded-FOP appliance), but for many basic situations (manager/admin as one simple example) lights on a phone are hard to beat. ___ 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 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] Opinions on renice or turning off swap or ramdis
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask opinions: Bear in mind I come from a FreeBSD background. Linux might behave differently. 1. asterisk -p == renice -20 ?? Why? If you have other things running on the machine, get a dedicated box for Asterisk. It might make sense to give it a mildly elevated priority, but running it at -20 might cause problems if you needed to get in and administer a runaway server. 2. I've turned off swap with no apparent ill effects. Can anyone commment on long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day) Unless Linux has a really poor swap strategy, this is a terrible idea. Even a mediocre swap implementation will begin swapping out lightly used pages when memory starts getting short. That swapping out actually *frees* memory up, memory needed by active processes. Turning off swap merely causes the system to work harder, and in the event the case where a lot of unexpected memory is being used, you're forced to keep it all in core - probably denying memory requests to processes that need them. What about when Asterisk has a really slow memory leak, growing a meg a day? In normal system design, while this is not desirable, it is simply swapped out to disk, and life goes on (at least for a lot longer than the without-swap case). Turn on swap. Turn on *big* swap. Set an alarm on swap so you're notified of any significant amount of paging. That's the best of all worlds. 3. Can anyone comment on using ramdisk as swap and whether this is a good idea or bad idea? RAMDISK as in something like a hardware RAMDISK? Go ahead, but you're throwing away money. Figuring out why a system with 4GB of memory and is only running Asterisk is swapping is a cheaper fix. A software RAMDISK? No way. You're eating up system RAM to provide for the lack of ... system RAM. Not smart. I'm using 2.6 kernel. I've modified the PCI latency in rc.local: setpci -v -s my T100P address latency_timer=ff Anyone else have any performance tips? Carefully profile your system to find out where the bottlenecks really are. Then get out the Attitude Adjuster (BOFH's find that it works nearly as well on systems as it does on people). Then go buy a system with none of those bottlenecks. ;-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Bothering with H323
Thanks Kido for the answer. I have not been able to make it work yet. My config files are: ;h323.conf [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=all dtmfmode=rfc2833 gatekeeper=200.123.148.17 AllowGKRouted=yes context=h323 [devgw] type=h323 e164=100 context=h323 ;extension.conf exten = 99,1,Dial(h323/) exten = 991112,1,Dial(h323/[EMAIL PROTECTED]) exten = 991113,1,Dial(h323/[EMAIL PROTECTED]/) I have tryed the three combination but no one works. I get this logs: -- Executing Dial(SIP/6386-76cc, h323/) in new stack -- Called == No one is available to answer at this time == Auto fallthrough, channel 'SIP/6386-76cc' status is 'NOANSWER' Could you help me again. How can I know if my Asterisk is registered on the remote GK? (it is a gnuGK machine with no access for me). Thank you very much... Nahuel Ramos. On Thu, 25 Nov 2004 03:21:48 -, kido noagbodji [EMAIL PROTECTED] wrote: Hi Nahuel, in you h323.conf file, add the following line gatekeeper = yougk.ipadress.here then create an asterisk endpoints in your gk like this [detgw] type=h323 e164=100 context=context Then if you h323 endpoint is registered and if you modify you extensions.conf file like this it should work exten = 99,1,Dial(h323/12345678) assuming that your h323 registered endpoints IPN(ANI) is 12345678. That should work. K. - Original Message - From: Nahuel Alejandro Ramos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, November 24, 2004 9:48 PM Subject: [Asterisk-Users] Bothering with H323 Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I route my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten = 99,1,Dial(h323/[EMAIL PROTECTED]) exten = 991112,1,Dial(h323/[EMAIL PROTECTED]/123456789) Please, someone could help me touching the h323.conf. Thank you very much... Nahuel Ramos. ___ 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 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] SIP Phones-Receptionist Setup
I would like you to name one PBX that does not support this behavior? Every system from Avaya including their Definity, Merlin Legend, Merlin Magix, Partner, and their new IP based PBXes support it, as do those from Mitel, Nortel, InteCom and every other system that I have ever used. A typical example is a manager/admin setup that works as follows: Partner is not a PBX, it is a key system. The Definity PBX does not directly provide key functionality. I can't speak to Merlin, not having used it myself. That said, Asterisk is a PBX like Definity, and should not support this. A FEP for Asterisk, that duplicates the functionality of a key system, should be developed, if it's in high enough demand. Like I said before, I am happy to spearhead the project development if anyone is interested. Greg ___ 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] Cannot open /dev/dsp
Norman Zhang wrote: Cannot open /dev/dsp: file or directory not found That means you probably don't have a soundcard configured. I don't have one in my test box either, but that doesn't prohibit asterisk from starting up. it just means you can't do certain things from the CLI. After stop chan_oss.so and chan_also.so, the above WARNING went away. You are right. I don't have a sound card in this box. It's suppose to be PBX. ALSA is started though. Try starting up asterisk in verbose mode, a-la: asterisk -vvvc [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe Also in /var/log/asterisk/messages, I see Unable to open pseudo channel for timing... Sound may be choppy. Unable to open IAX timing interface: No such file or directory Unable to get our IP address, Skinny disabled Appreciate if someone could give me a few pointers here. Regards, Norman Zhang ___ 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
