Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
In article <[EMAIL PROTECTED]>,
Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
>
> I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
>
> NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco
>
> Incomming calls from the telco to the asterisk box to the NEC work fine with
> indials and everything. Works sweet.
>
> Outbound from the NEC to the Asterisk box fail. Giving an long dial tone
> that then times out.
>
> Ie, pick up NEC handset, dial to get outside line, are given a dial tone,
> and then press numbers on keypad but dialtone continues and then eventually
> get an fast busy signal.
>
> Looking at the console, it doesn't seem like the * system is seeing the
> dialed number data from the NEC box. the reason I say this is that the
> asterisk box complains that it is accepting call from '' to ''
>
> Which just seems like no data is passing across.
>
> It is an NEC Xen Master that is configured to connect to an E1 and works
> fine. This E1 line has been replaced by a direct link to the * box to make
> the NEC box think the * box is the Telco â so I know this at least works.
> And I can also send a call into the NEC system fine.
>
> Does anyone know how to debug this? How can I see what data the NEC box is
> sending over the signaling channel in * ?
You need to understand how the NEC interacted with the Telco before the
Asterisk box was inserted.
For example, where is the dialtone generated? By the NEC or by the unit
connected to the NEC's port?
If the NEC were acting like an Asterisk box with a phone directly connected,
it would generate dialtone, accept all the DTMF digits and then send a call
request to the E1, complete with called-party number.
You description makes me wonder whether what is happening is that you
dial 9 or whatever on the NEC to get an "outside line", and it then
immediately connects through to the Asterisk box which gives dialtone,
and you then send DTMF digits in-band. Perhaps the Telco's E1 is set up
for this, but I'm not sure what is necessary for Asterisk to do in-band
DTMF dialling from an incoming E1 call. Perhaps someone else here could
comment.
You might want to try changing immediate=no to immediate=yes on span 1.
If that's the case, you might need a different dialplan too:
[from-nec]
exten => s,1,WaitExten
exten => _X.,1,Dial(Zap/g2/${EXTEN},,T)
In this example, the immediate=yes goes straight to the 's' extension,
at which you wait for the user to dial the outgoing number. You might
need to set the digit timeouts before doing WaitExten.
If that doesn't work, you could try doing this at the Asterisk
CLI> prompt:
pri set debug file /tmp/pri.txt
pri debug span 1
Then try making an outbound call, and then:
pri no debug span 1
pri unset debug file
Then post the contents of /tmp/pri.txt if it's not too huge, or else
put it up on a file server or web site and post the URL.
Cheers
Tony
> dahdi/system.conf looks like:
>
>
> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"
> # To NEC System
> span=1,0,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31
>
> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> # To E1
> span=2,1,0,ccs,hdb3,crc4
> bchan=32-46,48-62
> dchan=47
> echocanceller=mg2,32-46,48-62
>
> # Global data
>
> loadzone = au
> defaultzone = au
>
>
> asterisk/dahdi-channels.conf looks like:
>
>
> ; SPAN 1 connects to the NEC PBX system
> ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF
> group=1
> context=from-nec
> switchtype = euroisdn
> signalling = pri_net
> channel => 1-15,17-31
> immediate = no
> overlapdial = yes
>
> ; SPAN 2 connects to Telstra â 30 channel E1
> ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" B8ZS/ESF
> group=2
> context=from-pstn
> switchtype = euroisdn
> signalling = pri_cpe
> channel => 32-46,48-62
> immediate = no
> overlapdial = yes
> prindication = outofband
>
>
> asterisk/extensions.conf (relative bit) looks like:
>
>
> [from-pstn]
> exten => _73XX,1,Dial(DAHDI/g1/${EXTEN},,T)
>
> [from-nec]
> exten => _X.,1,Dial(DAHDI/g2/${EXTEN},,T)
> -=-=-=-=-=-
> [Alternative: text/html]
> -=-=-=-=-=-
> -=-=-=-=-=-
>
> ___
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>
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield <[EMAIL PROTECTED]
> wrote:
> In article <[EMAIL PROTECTED]>,
> Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
> > I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
> > NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco
> > Incomming calls from the telco to the asterisk box to the NEC work fine
> with
> > indials and everything. Works sweet.
> > Outbound from the NEC to the Asterisk box fail. Giving an long dial tone
> > that then times out.
>
> You need to understand how the NEC interacted with the Telco before the
> Asterisk box was inserted.
> You might want to try changing immediate=no to immediate=yes on span 1.
> If that's the case, you might need a different dialplan too:
> If that doesn't work, you could try doing this at the Asterisk
> CLI> prompt:
Yup, didn't work. Ended up timing out on the WaitExten command and looking
for the t exten, which means it didn't receive a number to dial...
Here is the verbose output:
-- Accepting call from '' to 's' on channel 0/31, span 1
-- Executing [EMAIL PROTECTED]:1] WaitExten("DAHDI/31-1", "") in new stack
[Nov 16 14:10:01] WARNING[7729]: pbx.c:7787 pbx_builtin_waitexten: Timeout
but no rule 't' in context 'from-nec'
== Spawn extension (from-nec, s, 1) exited non-zero on 'DAHDI/31-1'
-- Hungup 'DAHDI/31-1'
Because this didn't work, I changed it back to immediate = no.
And removed the WaitExten from the from-nec context.
Now I don't get the constant dial tone any more, I get an immediate busy
with asterisk reporting:
-- Extension 's' in context 'from-nec' from '' does not exist.
Rejecting call on channel 0/31, span 1
Which means it's not getting the number to dial somehow.
> Then post the contents of /tmp/pri.txt if it's not too huge, or else
> put it up on a file server or web site and post the URL.
>
Isn't too long. There are no other calls happening at the moment, so is
nice and short:
< Protocol Discriminator: Q.931 (8) len=19
< Call Ref: len= 2 (reference 1/0x1) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a3]
< Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
< Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 1/0x1) (Terminator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 81]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: Private network serving the local user (1)
> Ext: 1 Cause: Unallocated (unassigned) number (1), class
= Normal Event (0) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
Any ideas?
Mikel
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Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
On Sun, Nov 16, 2008 at 3:13 AM, Mikel Lindsaar <[EMAIL PROTECTED]> wrote: > On Sat, Nov 15, 2008 at 11:05 PM, Tony Mountifield < > [EMAIL PROTECTED]> wrote: > >> In article <[EMAIL PROTECTED]>, >> Mikel Lindsaar <[EMAIL PROTECTED]> wrote: >> > I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. >> > NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco >> > Incomming calls from the telco to the asterisk box to the NEC work fine >> with >> > indials and everything. Works sweet. >> > Outbound from the NEC to the Asterisk box fail. Giving an long dial >> tone >> > that then times out. >> > Isn't too long. There are no other calls happening at the moment, so is > nice and short: > Here is a pastie of: dahdi/system.conf http://www.pastie.org/315597 asterisk/chan_dahdi.conf http://www.pastie.org/315608 asterisk/dahdi_channels.confhttp://www.pastie.org/315612 Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
OK, made some progress.
With some help from friends on #Asterisk, found that the NEC was doing post
connect dialing.
So then, added the following to extensions.conf:
[from-nec]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Read(DialedNumber)
exten => s,n,Dial(DAHDI/g2/${DialedNumber},,T)
Which now captures the number and tries to dial, but then I get:
-- Accepting call from '' to 's' on channel 0/28, span 1
-- Executing [EMAIL PROTECTED]:1] Answer("DAHDI/28-1", "") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("DAHDI/28-1", "TIMEOUT(digit)=2") in
new
stack
-- Digit timeout set to 2
-- Executing [EMAIL PROTECTED]:3] Set("DAHDI/28-1", "TIMEOUT(response)=5")
in
new stack
-- Response timeout set to 5
-- Executing [EMAIL PROTECTED]:4] Read("DAHDI/28-1", "DialedNumber") in new
stack
-- User entered '1414040040'
-- Executing [EMAIL PROTECTED]:5] Dial("DAHDI/28-1",
"DAHDI/g2/1414040040,,T") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/1414040040
-- DAHDI/32-1 is proceeding passing it to DAHDI/28-1
-- Channel 0/1, span 2 got hangup request, cause 31
[Nov 16 16:09:30] WARNING[5828]: app_dial.c:827 wait_for_answer: Unable to
forward voice or dtmf
-- Hungup 'DAHDI/32-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/28-1' status is 'CHANUNAVAIL'
-- Channel 0/28, span 1 got hangup request, cause 16
-- Hungup 'DAHDI/28-1'
Ideas?
Mikel
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Re: [asterisk-users] Polycom low volume
Probably has nothing to do with Asterisk. You can set the volume and persistence in the phones config files. Michael On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee wrote: >Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't >figure out why the volume is so low. How can I adjust the volume control on >Asterisk? It's at max on the handset phones. > >Thanks! >Hin > > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and Queues
Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
Good. Got the darn thing working.
Problem was the NEC Xen Master does post-connect DTMF to dial. So I had to
read the digits after connect.
Then, I had to configure the PRI to be pridialplan = unknown
Thanks for your help Tony.
Here is the end result for Google's sake
#
### /etc/dahdi/system.conf
#
#
# Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Global data
loadzone = au
defaultzone = au
#
### /etc/asterisk/chan_dahdi.conf
#
[trunkgroups]
[channels]
#include dahdi-channels.conf
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
language=en
hidecallerid=no
callerid=asreceived
restrictcid=no
#
### /etc/asterisk/dahdi-channels.conf
#
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=1
context=from-nec
switchtype = euroisdn
signalling = pri_net
channel => 1-15,17-31
; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group = 2
switchtype = euroisdn
signalling = pri_cpe
pridialplan=unknown
context = from-pstn
channel=>32-46,48-62
#
### extensions.conf (relevant bits)
#
[from-pstn]
exten => _55XX,1,Dial(DAHDI/g1/${EXTEN},,T)
[from-nec]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Read(DialedNumber)
exten => s,n,Dial(DAHDI/g2/${DialedNumber},,Tr)
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Re: [asterisk-users] Polycom low volume
Actually, it could be within Asterisk, but only if you have Zaptel hardware. If you are only using SIP devices, then the problem is with the phone configuration. You really don't provide enough information to determine what is causing your problem. How are you provisioning the phones? What version of the SIP firmware is used on the phones? Are you calling from one phone to the other? Darrick Michael Graves wrote: > Probably has nothing to do with Asterisk. You can set the volume and > persistence in the phones config files. > > Michael > > On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee wrote: > >> Using a Polycom 550 and 650 phones on my Asterisk server for testing. I >> can't figure out why the volume is so low. How can I adjust the volume >> control on Asterisk? It's at max on the handset phones. >> >> Thanks! >> Hin >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > Michael Graves > mgravesmstvp.com > http://blog.mgraves.org > o713-861-4005 > c713-201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > fwd 54245 > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk GUI and SIP registration
I was playing with 1.6.0.1 and the latest gui and wondered how my sip did was registered after creating it? How does this take place, normally I made a register => command in sip.conf but don't see this in any files? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
In article <[EMAIL PROTECTED]>,
Mikel Lindsaar <[EMAIL PROTECTED]> wrote:
>
> Good. Got the darn thing working.
Good news! Been away from the computer for a while, so wasn't able to
keep up with your efforts.
> Problem was the NEC Xen Master does post-connect DTMF to dial. So I had to
> read the digits after connect.
>
> Then, I had to configure the PRI to be pridialplan = unknown
>
> Thanks for your help Tony.
You're welcome. Glad to have provided some pointers.
Actually, if Read() works, then WaitExten should have worked too. I expect
what was missing was the Answer(). So this ought to work as an alternative:
[from-nec]
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,WaitExten()
exten => _X.,1,Dial(DAHDI/g2/${EXTEN},,Tr)
This would allow you to match different patters to route to different
destinations (e.g. some calls to VoIP or to internal functions such as
conferencing).
Also, there is one improvement I think might be made to either, and that
is to replace the 'r' flag (always generate ringing) with a call to
Progress() before the Dial:
exten => _X.,1,Progress()
exten => _X.,n,Dial(DAHDI/g2/${EXTEN},,T)
or similarly for your Read() solution. Unless I've missed something, that
ought to allow your callers to hear proper busy tones if the call is busy,
or ringing tone if it rings. It might even be possible to put Progress()
before WaitExten() to avoid duplicating it in different dial patterns.
Cheers
Tony
> Here is the end result for Google's sake
>
>
>
> #
> ### /etc/dahdi/system.conf
> #
> #
> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31
>
> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> span=2,2,0,ccs,hdb3,crc4
> # termtype: te
> bchan=32-46,48-62
> dchan=47
> echocanceller=mg2,32-46,48-62
>
>
> # Global data
>
> loadzone = au
> defaultzone = au
>
>
>
>
> #
> ### /etc/asterisk/chan_dahdi.conf
> #
> [trunkgroups]
>
> [channels]
>
> #include dahdi-channels.conf
>
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
>
>
> callgroup=1
> pickupgroup=1
>
> language=en
> hidecallerid=no
> callerid=asreceived
> restrictcid=no
>
>
>
>
>
>
> #
> ### /etc/asterisk/dahdi-channels.conf
> #
> ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
> group=1
> context=from-nec
> switchtype = euroisdn
> signalling = pri_net
> channel => 1-15,17-31
>
> ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> group = 2
> switchtype = euroisdn
> signalling = pri_cpe
> pridialplan=unknown
> context = from-pstn
> channel=>32-46,48-62
>
>
>
>
>
> #
> ### extensions.conf (relevant bits)
> #
>
> [from-pstn]
> exten => _55XX,1,Dial(DAHDI/g1/${EXTEN},,T)
>
> [from-nec]
> exten => s,1,Answer()
> exten => s,n,Set(TIMEOUT(digit)=2)
> exten => s,n,Set(TIMEOUT(response)=5)
> exten => s,n,Read(DialedNumber)
> exten => s,n,Dial(DAHDI/g2/${DialedNumber},,Tr)
>
> -=-=-=-=-=-
> [Alternative: text/html]
> -=-=-=-=-=-
> -=-=-=-=-=-
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> -=-=-=-=-=-
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] IAX2 client for "eee pc 1000"
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and not fully available in all distros. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RV: MixMonitor and Queues
Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
Joseph wrote: > What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux > software)? > > I'll eventually replace this crippled Linux with something better but I don't > time to play around with it as most divers and modules are still too new and > not fully available in all distros. > > Kiax2 works pretty good. http://www.forschung-direkt.eu/ -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI and SIP registration
This would be better on the Asterisk-gui list, but it's because it's written in users.conf registersip=yes -bk - Original Message - From: "Joseph L. Casale" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, November 15, 2008 4:18:25 PM GMT -06:00 Central America Subject: [asterisk-users] Asterisk GUI and SIP registration I was playing with 1.6.0.1 and the latest gui and wondered how my sip did was registered after creating it? How does this take place, normally I made a register => command in sip.conf but don't see this in any files? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
On 11/15/08 18:04, David wrote: >Joseph wrote: >> What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux >> software)? >> >> I'll eventually replace this crippled Linux with something better but I >> don't time to play around with it as most divers and modules are still too >> new and >> not fully available in all distros. >> >> >Kiax2 works pretty good. >http://www.forschung-direkt.eu/ It keeps complaining about /lib/tls/libc.so.6 'GLIBC_2.4' not found. How do you install this library on EEE pc Xandros? (I know Xandros is Debian based) but this is eee pc. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
Joseph wrote: > > It keeps complaining about /lib/tls/libc.so.6 'GLIBC_2.4' not found. > > How do you install this library on EEE pc Xandros? (I know Xandros is Debian > based) but this is eee pc. > > You should ask on another list but this should get you started; http://forum.eeeuser.com/viewtopic.php?id=875 -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
Hi Joseph, Not directly related to your question (it's more an answer for the "something better" part of your plan), but I've loaded Ubuntu onto my Asus eeePC 4G Surf, and I've found that ZoIPer works pretty well. Cheers, AR -- Alex Robar [EMAIL PROTECTED] On Sat, Nov 15, 2008 at 5:49 PM, Joseph <[EMAIL PROTECTED]> wrote: > What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux > software)? > > I'll eventually replace this crippled Linux with something better but I > don't time to play around with it as most divers and modules are still too > new and > not fully available in all distros. > > -- > #Joseph > GPG KeyID: ED0E1FB7 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
Well, I've tried to find Ubuntu but so fare I'm not sure which one. I have eee pc 1000 (one with 40GB SSD so plenty of room for any modern distro. Which ubuntu did you loaded? It has to be something that loads onto USB bootable stick (and not through Windows as I don't have one). -- #Joseph GPG KeyID: ED0E1FB7 On 11/15/08 19:01, Alex Robar wrote: >Hi Joseph, > >Not directly related to your question (it's more an answer for the >"something better" part of your plan), but I've loaded Ubuntu onto my Asus >eeePC 4G Surf, and I've found that ZoIPer works pretty well. > >Cheers, >AR > >-- >Alex Robar >[EMAIL PROTECTED] > > >On Sat, Nov 15, 2008 at 5:49 PM, Joseph <[EMAIL PROTECTED]> wrote: > >> What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux >> software)? >> >> I'll eventually replace this crippled Linux with something better but I >> don't time to play around with it as most divers and modules are still too >> new and >> not fully available in all distros. >> >> -- >> #Joseph >> GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX -> PRI -> * -> Telco not working
On Sun, Nov 16, 2008 at 9:37 AM, Tony Mountifield <[EMAIL PROTECTED]>wrote: > > Actually, if Read() works, then WaitExten should have worked too. I expect > what was missing was the Answer(). So this ought to work as an alternative: It doesn't for some reason... I end up getting the timeout call [Nov 16 23:22:51] WARNING[8213]: pbx.c:7787 pbx_builtin_waitexten: Timeout but no rule 't' in context 'from-nec' == Spawn extension (from-nec, s, 3) exited non-zero on 'DAHDI/28-1' -- Hungup 'DAHDI/28-1' Also, there is one improvement I think might be made to either, and that > is to replace the 'r' flag (always generate ringing) with a call to > Progress() before the Dial: I'll give that a shot And ideas why the waitexten is not getting the digits? I am getting another weird error as well that might shed some light. Calling from the NEC phone digital handsets works 100%. Calling from the telco and into the NEC system also works fine. But when I try to call out using a POTS phone connected to an analog line on the NEC, it seems like the numbers get mangled on either their way to Asterisk (unlikely as the NEC was making calls out the E1 from POTS lines yesterday) or from the translation that the asterisk box is doing to talk to the NEC. For example, I dial: 9692 the NEC system prepends it with 1414 (which is our telco access prefix) and then dials. If I do this from an NEC digital extension I get 14149692, but if I do it from an NEC POTS extension I get 1942124000 Strange :) Incomming from Telco => * => NEC => POTS line works fine. Wrap your brain cells around that one :) Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
The solution for the problem of an IAX client is a SIP client. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
I installed eeeUbuntu ( http://www.ubuntu-eee.com/ ). I followed instructions on their wiki to create the USB-stick installer from another Ubuntu PC that I have. It looks like they've made the process even easier now, with a GUI-based application that will prep a USB stick with any ISO that you choose. Once you've got Ubuntu running, the 1000 series should have more than enough horsepower to run any number of great IAX2 softphones. As I mentioned, I'm running ZoIPer, but it's not the most light or stable application out there. As David suggested, KIAX2 is pretty good. If SIP works as an alternative, there's a plethora of other great applications. -- Alex Robar [EMAIL PROTECTED] On Sat, Nov 15, 2008 at 8:13 PM, Joseph <[EMAIL PROTECTED]> wrote: > Well, I've tried to find Ubuntu but so fare I'm not sure which one. I have > eee pc 1000 (one with 40GB SSD so plenty of room for any modern distro. > Which ubuntu did you loaded? It has to be something that loads onto USB > bootable stick (and not through Windows as I don't have one). > > -- > #Joseph > GPG KeyID: ED0E1FB7 > > On 11/15/08 19:01, Alex Robar wrote: > >Hi Joseph, > > > >Not directly related to your question (it's more an answer for the > >"something better" part of your plan), but I've loaded Ubuntu onto my Asus > >eeePC 4G Surf, and I've found that ZoIPer works pretty well. > > > >Cheers, > >AR > > > >-- > >Alex Robar > >[EMAIL PROTECTED] > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom low volume
Here are more information as requested: Asterisk v. 1.4 (running PBX in a Flash) Using Zaptel, TDM800P card Polycom running: 3.03 SIP Firmware Provisioning by: FTP I am calling from my Polycom to other land line phones. Hope I provided enough information. Thanks! Hin --- On Sat, 11/15/08, Darrick Hartman <[EMAIL PROTECTED]> wrote: > From: Darrick Hartman <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Polycom low volume > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Saturday, November 15, 2008, 1:44 PM > Actually, it could be within Asterisk, but only if you have > Zaptel > hardware. If you are only using SIP devices, then the > problem is with > the phone configuration. You really don't provide > enough information to > determine what is causing your problem. How are you > provisioning the > phones? What version of the SIP firmware is used on the > phones? Are > you calling from one phone to the other? > > Darrick > > Michael Graves wrote: > > Probably has nothing to do with Asterisk. You can set > the volume and > > persistence in the phones config files. > > > > Michael > > > > On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee > wrote: > > > >> Using a Polycom 550 and 650 phones on my Asterisk > server for testing. I can't figure out why the volume > is so low. How can I adjust the volume control on Asterisk? > It's at max on the handset phones. > >> > >> Thanks! > >> Hin > >> > >> > >> > >> > >> ___ > >> -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > -- > > Michael Graves > > mgravesmstvp.com > > http://blog.mgraves.org > > o713-861-4005 > > c713-201-1262 > > sip:[EMAIL PROTECTED] > > skype mjgraves > > fwd 54245 > > > > > > > > > > ___ > > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom low volume
At 21:06 11/15/2008, hin lee wrote: >Here are more information as requested: > >Asterisk v. 1.4 (running PBX in a Flash) >Using Zaptel, TDM800P card >Polycom running: 3.03 SIP Firmware >Provisioning by: FTP > >I am calling from my Polycom to other land line phones. Hope I >provided enough information. Why don't you post a link to your sip.cfg? Typical PhoneXX.cfg? > >Thanks! >Hin > > >--- On Sat, 11/15/08, Darrick Hartman <[EMAIL PROTECTED]> wrote: > >> From: Darrick Hartman <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-users] Polycom low volume >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >> Date: Saturday, November 15, 2008, 1:44 PM >> Actually, it could be within Asterisk, but only if you have >> Zaptel >> hardware. If you are only using SIP devices, then the >> problem is with >> the phone configuration. You really don't provide >> enough information to >> determine what is causing your problem. How are you >> provisioning the >> phones? What version of the SIP firmware is used on the >> phones? Are >> you calling from one phone to the other? >> >> Darrick >> >> Michael Graves wrote: >> > Probably has nothing to do with Asterisk. You can set >> the volume and >> > persistence in the phones config files. >> > >> > Michael >> > >> > On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee >> wrote: >> > >> >> Using a Polycom 550 and 650 phones on my Asterisk >> server for testing. I can't figure out why the volume >> is so low. How can I adjust the volume control on Asterisk? >> It's at max on the handset phones. >> >> >> >> Thanks! >> >> Hin >> >> >> >> >> >> >> >> >> >> ___ >> >> -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> >> >> >> asterisk-users mailing list >> >> To UNSUBSCRIBE or update options visit: >> >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > >> > -- >> > Michael Graves >> > mgravesmstvp.com >> > http://blog.mgraves.org >> > o713-861-4005 >> > c713-201-1262 >> > sip:[EMAIL PROTECTED] >> > skype mjgraves >> > fwd 54245 >> > >> > >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
Alex Balashov wrote: > The solution for the problem of an IAX client is a SIP client. > That's not a particularly good solution if you have a NAT between your client and Asterisk. IAX is still *much* easier to get working through a firewall. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 client for "eee pc 1000"
I just tried it from a stick, nice looking but too many bugs: camera and mic don't work: so useless for conversation. https://answers.launchpad.net/ubuntu-eee/+question/43167 and that was posted 2008-08-26 now is Nov. 15/08 and it still doesn't work, so no mic no conversation :-/ Some suggested SIP; no SIP is not a solution if I'm behind a firewall especially on someone's network. -- #Joseph GPG KeyID: ED0E1FB7 On 11/15/08 20:49, Alex Robar wrote: >I installed eeeUbuntu ( http://www.ubuntu-eee.com/ ). I followed >instructions on their wiki to create the USB-stick installer from another >Ubuntu PC that I have. It looks like they've made the process even easier >now, with a GUI-based application that will prep a USB stick with any ISO >that you choose. Once you've got Ubuntu running, the 1000 series should have >more than enough horsepower to run any number of great IAX2 softphones. As I >mentioned, I'm running ZoIPer, but it's not the most light or stable >application out there. As David suggested, KIAX2 is pretty good. If SIP >works as an alternative, there's a plethora of other great applications. > >-- >Alex Robar >[EMAIL PROTECTED] > > >On Sat, Nov 15, 2008 at 8:13 PM, Joseph <[EMAIL PROTECTED]> wrote: > >> Well, I've tried to find Ubuntu but so fare I'm not sure which one. I have >> eee pc 1000 (one with 40GB SSD so plenty of room for any modern distro. >> Which ubuntu did you loaded? It has to be something that loads onto USB >> bootable stick (and not through Windows as I don't have one). >> >> -- >> #Joseph >> GPG KeyID: ED0E1FB7 >> >> On 11/15/08 19:01, Alex Robar wrote: >> >Hi Joseph, >> > >> >Not directly related to your question (it's more an answer for the >> >"something better" part of your plan), but I've loaded Ubuntu onto my Asus >> >eeePC 4G Surf, and I've found that ZoIPer works pretty well. >> > >> >Cheers, >> >AR >> > >> >-- >> >Alex Robar >> >[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom low volume
Links to my configuration files for the polycom phone. As you'll see, majority of my settings are default. Hope it will help you to determine where my problem is at. MAC Address cfg file http://docs.google.com/Doc?id=dggrkn86_2dc3qfdgr&hl=en Extension cfg file http://docs.google.com/Doc?id=dggrkn86_5ss94tkf6&hl=en Phone cfg file http://docs.google.com/Doc?id=dggrkn86_4csdzthf9&hl=en Server cfg file http://docs.google.com/Doc?id=dggrkn86_6gp7hr9fg&hl=en SIP cfg file http://docs.google.com/Doc?id=dggrkn86_3djzb86d7&hl=en --- On Sat, 11/15/08, hin lee <[EMAIL PROTECTED]> wrote: > From: hin lee <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Polycom low volume > To: "Doug" <[EMAIL PROTECTED]>, "Asterisk Users" > > Date: Saturday, November 15, 2008, 10:40 PM > Attached, my configuration files for the polycom phone. As > you'll see, majority of my settings are default. Hope > it will help you to determine where my problem is at. > > Thanks! > Hin > > --- On Sat, 11/15/08, Doug <[EMAIL PROTECTED]> wrote: > > > From: Doug <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] Polycom low volume > > To: [EMAIL PROTECTED], asterisk-users@lists.digium.com > > Date: Saturday, November 15, 2008, 7:20 PM > > At 21:06 11/15/2008, hin lee wrote: > > >Here are more information as requested: > > > > > >Asterisk v. 1.4 (running PBX in a Flash) > > >Using Zaptel, TDM800P card > > >Polycom running: 3.03 SIP Firmware > > >Provisioning by: FTP > > > > > >I am calling from my Polycom to other land line > phones. > > Hope I > > >provided enough information. > > > > Why don't you post a link to your sip.cfg? > > > > Typical PhoneXX.cfg? > > > > > > > > > >Thanks! > > >Hin > > > > > > > > >--- On Sat, 11/15/08, Darrick Hartman > > <[EMAIL PROTECTED]> wrote: > > > > > >> From: Darrick Hartman > > <[EMAIL PROTECTED]> > > >> Subject: Re: [asterisk-users] Polycom low > volume > > >> To: "Asterisk Users Mailing List - > > Non-Commercial Discussion" > > > > > >> Date: Saturday, November 15, 2008, 1:44 PM > > >> Actually, it could be within Asterisk, but > only if > > you have > > >> Zaptel > > >> hardware. If you are only using SIP devices, > then > > the > > >> problem is with > > >> the phone configuration. You really > don't > > provide > > >> enough information to > > >> determine what is causing your problem. How > are > > you > > >> provisioning the > > >> phones? What version of the SIP firmware is > used > > on the > > >> phones? Are > > >> you calling from one phone to the other? > > >> > > >> Darrick > > >> > > >> Michael Graves wrote: > > >> > Probably has nothing to do with > Asterisk. You > > can set > > >> the volume and > > >> > persistence in the phones config files. > > >> > > > >> > Michael > > >> > > > >> > On Fri, 14 Nov 2008 22:43:45 -0800 > (PST), hin > > lee > > >> wrote: > > >> > > > >> >> Using a Polycom 550 and 650 phones > on my > > Asterisk > > >> server for testing. I can't figure out > why > > the volume > > >> is so low. How can I adjust the volume > control on > > Asterisk? > > >> It's at max on the handset phones. > > >> >> > > >> >> Thanks! > > >> >> Hin > > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > > ___ > > >> >> -- Bandwidth and Colocation Provided > by > > >> http://www.api-digital.com -- > > >> >> > > >> >> asterisk-users mailing list > > >> >> To UNSUBSCRIBE or update options > visit: > > >> >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >> >> > > >> > > > >> > -- > > >> > Michael Graves > > >> > mgravesmstvp.com > > >> > http://blog.mgraves.org > > >> > o713-861-4005 > > >> > c713-201-1262 > > >> > sip:[EMAIL PROTECTED] > > >> > skype mjgraves > > >> > fwd 54245 > > >> > > > >> > > > >> > > > >> > > > >> > > > ___ > > >> > -- Bandwidth and Colocation Provided by > > >> http://www.api-digital.com -- > > >> > > > >> > asterisk-users mailing list > > >> > To UNSUBSCRIBE or update options visit: > > >> > > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >> > > >> > > >> > ___ > > >> -- Bandwidth and Colocation Provided by > > >> http://www.api-digital.com -- > > >> > > >> asterisk-users mailing list > > >> To UNSUBSCRIBE or update options visit: > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > >___ > > >-- Bandwidth and Colocation Provided by > > http://www.api-digital.com -- > > > > > >asterisk-users mailing list > > >To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
