date:20050306

On Asterisk 1.0 with a 4-port Digium FXO card, voicemails from the PSTN 
have volume so low they often can't be heard. Worse, callers sometimes get 
cut off in the middle of leaving a message. It is extremely frustrating to 
hear ...and my number is...END OF MESSAGE

A search of the archives shows this is known bug:
http://bugs.digium.com/bug_view_page.php?bug_id=0002023.
I'm relatively new to * and don't know what parameters I can tweak to fix 
this.

For example, where does pstnVMgain=5 go?
And are there other parameters I can use to fix this problem?
thanks
dn
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[Asterisk-Users] Need help on * anf HFC.

2005-03-06 Thread Ramon Roca

Hi, I'm a newbie on * trying to setup an HFC card.
I'm locked for many days getting the all-circuits-busy. And no idea what 
else to look for/how to diagnose.
I'm in Spain, I've tried changing many parameters on zapata/zaptelcong 
with no luck, also NT  TE modes (honsetly, I've no idea what is).
Any clue will be very much appreciated!

I've installed [EMAIL PROTECTED] on my RH9, and on top of that, bristuff-0.2.0-RC7f 
(that reinstalls asterisk).

Here you have what I've:
lspci output:
01:02.0 Network controller: Cologne Chip Designs GmbH ISDN network 
controller [HFC-PCI] (rev 02)

When loading zaptel drivers:
Mar  6 21:29:13 linux-1 kernel: Zapata Telephony Interface Registered on 
major 196
Mar  6 21:29:13 linux-1 kernel: PCI: Enabling device 01:02.0 ( - 0003)
Mar  6 21:29:13 linux-1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0 
configured at mem 0xf91c5e00 fifo 0xf7598000(0x37598000) IRQ 5 HZ 100
Mar  6 21:29:13 linux-1 kernel: zaphfc: Card 0 configured for NT mode
Mar  6 21:29:13 linux-1 kernel: zaphfc: 1 hfc-pci card(s) in this box.
Mar  6 21:29:13 linux-1 kernel: Registered tone zone 3 (Netherlands)
mar  6 21:29:13 linux-1 zaptel: Loading zaptel framework:  succeeded
Mar  6 21:29:15 linux-1 kernel: Specify address with base=0xN
Mar  6 21:29:15 linux-1 kernel: Registered Tormenta2 PCI
Mar  6 21:29:17 linux-1 kernel: Registered tone zone 3 (Netherlands)
mar  6 21:29:17 linux-1 zaptel: Running ztcfg:  succeeded
mar  6 21:29:34 linux-1 su(pam_unix)[21409]: session opened for user 
asterisk by (uid=0)
mar  6 21:29:34 linux-1 su(pam_unix)[21409]: session closed for user 
asterisk
mar  6 21:29:40 linux-1 su(pam_unix)[21484]: session opened for user 
asterisk by (uid=0)
mar  6 21:29:40 linux-1 su(pam_unix)[21484]: session closed for user 
asterisk
Mar  6 21:30:01 linux-1 kernel: zaphfc: bchan rx fifo not enough bytes 
to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun.
Mar  6 21:30:49 linux-1 kernel: zaphfc: bchan rx fifo not enough bytes 
to receive! (z1=528, z2=527, wanted 8 got 2), probably a buffer overrun.
Mar  6 21:32:26 linux-1 last message repeated 2 times
Mar  6 21:34:03 linux-1 last message repeated 2 times
Mar  6 21:35:40 linux-1 last message repeated 2 times

My /etc/zaptel.conf:
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
My zapata.conf:
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode
;signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
signalling = bri_net
pridialplan = local
;prilocaldialplan = local
; nationalprefix = 0
;internationalprefix = 00
; trust user provided callerid (clip no screening)?
;pritrustusercid = no
echocancel=yes
;echotraining = 100
;echocancelwhenbridged=yes
immediate=yes
group = 1
context=outbound-trunks
channel = 1-2
Asterisk console while trying to use the dial out trunk:
Mar  6 21:40:01 DEBUG[21452]: Setting NAT on RTP to 0
Mar  6 21:40:01 DEBUG[21452]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Response 101: Found
Mar  6 21:40:01 DEBUG[21452]: Setting NAT on RTP to 0
Mar  6 21:40:01 DEBUG[21452]: Check for res for 200
Mar  6 21:40:01 DEBUG[21452]: Call from user '200' is 1 out of 0
Mar  6 21:40:01 DEBUG[21452]: build_route: Contact hop: Roser Roca 
sip:[EMAIL PROTECTED]:5061
Mar  6 21:40:01 VERBOSE[21452]: -- Executing Macro(SIP/200-1cf6, 
dialout-default|9639712471) in new stack
Mar  6 21:40:01 WARNING[21452]: ast_yyerror(): syntax error: parse 
error; Input:
fooEl Serrat = foo
^
^
Mar  6 21:40:01 DEBUG[21452]: Expression is 'fooEl'
Mar  6 21:40:01 VERBOSE[21452]: -- Executing GotoIf(SIP/200-1cf6, 
fooEl?4) in new stack
Mar  6 21:40:01 DEBUG[21452]: Not taking any branch
Mar  6 21:40:01 VERBOSE[21452]: -- Executing 
SetCallerID(SIP/200-1cf6, El Serrat) in new stack
Mar  6 21:40:01 VERBOSE[21452]: -- Executing Goto(SIP/200-1cf6, 
6) in new stack
Mar  6 21:40:01 VERBOSE[21452]: -- Goto (macro-dialout-default,s,6)
Mar  6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6, 
ZAP/g0/9639712471) in new stack
Mar  6 21:40:01 NOTICE[21452]: Unable to create channel of type 'ZAP'
Mar  6 21:40:01 VERBOSE[21452]:   == Everyone is busy/congested at this time
Mar  6 21:40:01 DEBUG[21452]: Exiting with DIALSTATUS=CHANUNAVAIL.
Mar  6 21:40:01 VERBOSE[21452]: -- Executing Macro(SIP/200-1cf6, 
outisbusy) in new stack
Mar  6 21:40:01 VERBOSE[21452]: -- Executing 
Playback(SIP/200-1cf6, allison7/all-circuits-busy-now) in new stack
Mar  6 21:40:01 DEBUG[21452]: Ooh, format changed from unknown to ulaw
Mar  6 21:40:01 DEBUG[21452]: Scheduling timer at 160 sample intervals
Mar  6 21:40:01 VERBOSE[21452]: -- Playing 
'allison7/all-circuits-busy-now' (language 'en')
Mar  6 21:40:01 DEBUG[21452]: Stopping

[Asterisk-Users] SpanDSP: Training failed (sequence failed)

2005-03-06 Thread CClarke

Hello All ~
Having problems sending and receiving faxes with SpanDSP. I am testing on a
simple 2 analog POTS to 2x X100p set up, connecting one line to a Konica 720
fax machine to test, or with other remote fax machines. Voice calls are
working pretty well now. Platform is P3/800MHz/256MB/FC1.

* recognizes faxes, and passes calls to RxFax and TxFax OK, but 99.9% of the
time no fax content is sent or received. (On one occasion coincidentally a
junk fax came in successfully while I was testing!).

I've reviewed previous posts and but can't find any relevant advice on where
to go next, since the fax negotiation seems to go OK so far, but then die (see
below), and no content is transmitted. The final error message from RxFax is:
Training failed (sequence failed)

Would really appreciate some expert advice on what this means and how to
fix...
Christina.

PS: Previous somewhat relevant posts I've been able to find:
http://lists.digium.com/pipermail/asterisk-users/2004-June/051143.html
http://lists.digium.com/pipermail/asterisk-users/2005-February/090978.html

Here's versions I'm using:
asterisk CVS-v1-0-02/20/05-17:04:48
spandsp 0.0.1k
libtiff 3.5.7

Here's the relevant CLI output:
-- Redirecting Zap/1-1 to fax extension
  == Spawn extension (default, fax, 0) exited non-zero on 'Zap/1-1'
-- Executing SetVar(Zap/1-1,
FAXFILE=/var/spool/asterisk-fax/1110134598.7.tif) in new stack
-- Executing RxFAX(Zap/1-1, /var/spool/asterisk-fax/1110134598.7.tif)
in new stack
Changed from phase 0 to 1
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Slow carrier up
Slow carrier down
Start receiving document
Changed from phase 1 to 4
Sending ident
 CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
 DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
Slow carrier up
Slow carrier down
Slow carrier up
 TSI: 43 32 31 32 31 35 35 35 39 31 35 20 20 20 20 20 20 20 20 20 20
TSI without final frame tag
Remote fax gave TSI as: 5195551212
 DCS: 83 00 06 00
DCS with final frame tag
In state 9
DCS:
Can receive fax
Selected data signalling rate: V.29, 9600bps
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 20ms
Get at 9600
Changed from phase 3 to 5
Fast carrier up
Coarse carrier frequency 1741.58 (6)
Fast carrier down
Fast carrier up
Coarse carrier frequency 1700.10 (72)
Training error 216.035327
Training failed (convergence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Training failed (sequence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Coarse carrier frequency 1699.92 (72)
Training error 216.181752
Training failed (convergence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Training failed (sequence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Coarse carrier frequency 1699.77 (72)
Training error 236.605116
Training failed (convergence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Training failed (sequence failed)
Fast carrier training failed
Fast carrier down
Fast carrier up
Coarse carrier frequency 1759.74 (4)
Fast carrier down
Fast carrier up
Fast carrier down
-- Hungup 'Zap/1-1'

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[Asterisk-Users] SNMP and Astersik

2005-03-06 Thread Anderson Alves de Albuquerque


 I have FXO (DIGIUM) with Asterisk (PBX). How can I use SNMP in Asterisk 
to access FXO?

 I need to known if FXO has the LINE with PSTN free to new phone call. Is 
this possible? How?





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Re: [Asterisk-Users] Need help on * anf HFC.

2005-03-06 Thread Julian J. M.

Hello,

I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context from-pstn if you want to receive calls.

group = 0
context=from-pstn
channel = 1-2

BTW, i'm from Spaintoo, and I'm really interested in knowing if your
setup works ;)

On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca [EMAIL PROTECTED] wrote:
 [channels]
 group = 1
 context=outbound-trunks
 channel = 1-2


 Mar  6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6,
 ZAP/g0/9639712471) in new stack

g0 means channel group 0, and you had group 1


Julian.
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RE: [Asterisk-Users] voicemail volume

2005-03-06 Thread Marty Mastera

 
 On Asterisk 1.0 with a 4-port Digium FXO card, voicemails 
 from the PSTN have volume so low they often can't be heard. 
 Worse, callers sometimes get cut off in the middle of leaving 
 a message. It is extremely frustrating to hear ...and my 
 number is...END OF MESSAGE
 
 A search of the archives shows this is known bug:
 
 http://bugs.digium.com/bug_view_page.php?bug_id=0002023.
 
 I'm relatively new to * and don't know what parameters I can 
 tweak to fix this.
 
 For example, where does pstnVMgain=5 go?
 
 And are there other parameters I can use to fix this problem?
 
 thanks
 
 dn

The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).

Marty
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Re: [Asterisk-Users] Need help on * anf HFC.

2005-03-06 Thread Ramon Roca

Hey Julian, thanks! It really make a difference. Thanks for pointing me 
to this. Stupid newbie mistake.
Yes, I'm using AMP, it was bundled with [EMAIL PROTECTED]
Now I'm not longer getting the all-the-circuits-are-busy-now, but still 
doesn't dial out, now I'm getting the congestion tone.
Maybe I'm loading the zaphfc with wrong parameters for Spanish ISDN?

I'm just using a regular ISDN at home, and plugged the RJ45 cable at the 
same port where was the Euromix RDSI phone.

Here it is the current  * console while dialing out:
Mar  6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0
Mar  6 22:44:58 DEBUG[3700]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Response 101: Found
Mar  6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0
Mar  6 22:44:58 DEBUG[3700]: Check for res for 200
Mar  6 22:44:58 DEBUG[3700]: Call from user '200' is 1 out of 0
Mar  6 22:44:58 DEBUG[3700]: build_route: Contact hop: Roser Roca 
sip:[EMAIL PROTECTED]:5061
Mar  6 22:44:58 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90, 
dialout-default|9639712471) in new stack
Mar  6 22:44:58 WARNING[3700]: ast_yyerror(): syntax error: parse error; 
Input:
fooEl Serrat = foo
^
^
Mar  6 22:44:58 DEBUG[3700]: Expression is 'fooEl'
Mar  6 22:44:58 VERBOSE[3700]: -- Executing GotoIf(SIP/200-bd90, 
fooEl?4) in new stack
Mar  6 22:44:58 DEBUG[3700]: Not taking any branch
Mar  6 22:44:58 VERBOSE[3700]: -- Executing 
SetCallerID(SIP/200-bd90, El Serrat) in new stack
Mar  6 22:44:58 VERBOSE[3700]: -- Executing Goto(SIP/200-bd90, 
6) in new stack
Mar  6 22:44:58 VERBOSE[3700]: -- Goto (macro-dialout-default,s,6)
Mar  6 22:44:58 VERBOSE[3700]: -- Executing Dial(SIP/200-bd90, 
ZAP/g0/9639712471) in new stack
Mar  6 22:44:58 VERBOSE[3700]: -- Called g0/9639712471
Mar  6 22:45:02 VERBOSE[3700]: -- Channel 0/1, span 1 got hangup
Mar  6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Mar  6 22:45:02 DEBUG[3700]: Hangup: channel: 1 index = 0, normal = 15, 
callwait = -1, thirdcall = -1
Mar  6 22:45:02 DEBUG[3700]: Already hungup...  Calling hangup once, and 
clearing call
Mar  6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1
Mar  6 22:45:02 DEBUG[3700]: Set option TDD MODE, value: OFF(0) on Zap/1-1
Mar  6 22:45:02 DEBUG[3700]: Updated conferencing on 1, with 0 
conference users
Mar  6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Mar  6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1
Mar  6 22:45:02 VERBOSE[3700]: -- Hungup 'Zap/1-1'
Mar  6 22:45:02 VERBOSE[3700]:   == No one is available to answer at 
this time
Mar  6 22:45:02 DEBUG[3700]: Exiting with DIALSTATUS=NOANSWER.
Mar  6 22:45:02 VERBOSE[3700]: -- Executing 
Congestion(SIP/200-bd90, ) in new stack
Mar  6 22:45:02 VERBOSE[3700]:   == Spawn extension 
(macro-dialout-default, s, 7) exited non-zero on 'SIP/200-bd90' in macro 
'dialout-default'
Mar  6 22:45:02 VERBOSE[3700]:   == Spawn extension (from-internal, 
9639712471, 1) exited non-zero on 'SIP/200-bd90'
Mar  6 22:45:02 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90, 
hangupcall) in new stack


En/na Julian J. M. ha escrit:
Hello,
I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context from-pstn if you want to receive calls.
group = 0
context=from-pstn
channel = 1-2
BTW, i'm from Spaintoo, and I'm really interested in knowing if your
setup works ;)
On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca [EMAIL PROTECTED] wrote:
 

[channels]
group = 1
context=outbound-trunks
channel = 1-2
   


 

Mar  6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6,
ZAP/g0/9639712471) in new stack
   

g0 means channel group 0, and you had group 1
Julian.
 

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RE: [Asterisk-Users] voicemail volume

On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN - * voicemail?
thanks
dn
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RE: [Asterisk-Users] voicemail volume

2005-03-06 Thread Marty Mastera


  The full text of the bug you reference above indicates that 
 pstnVMgain
  was (or is) part of an ongoing feature request/bug report 
 and has not 
  been implemented for use at this time (and may never be).
 
 Right. So -- what can I do to boost volume of PSTN - * voicemail?
 
 thanks
 
 dn

The only way I've been able to do it so far is to edit zapata.conf and
play with the rxgain= setting (raising it from zero) until the incoming
audio is loud enough in the resulting vm recordings. 
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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Alistair Cunningham

Thomas,
The definitive guide of what versions can be upgraded to what is at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html
In particular, look at tables 2 and 3.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
Thomas Trepper wrote:
Hi all,
i am new to this list and i dot not know, if anybody had already the 
same problem. I have two cisco 7960 which i want to upgrade to sip. Has 
somebody already taken the upgrade-process for special hints and 
suggestions? I have already visited the cisco-page and i have read the 
proposal for the migration. Is there a special order of firmware-upgrades?

Thanks a lot
Thomas
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[Asterisk-Users] Trying to get 2 SIP phones to work

Im new to Astererisk. I compiled the latest CVS and setup the server. It 
looks like things are working. I'm running kphone, x-lite and sjphone to 
test things out.  The kphone (local to the asterisk server) can call and 
receive calls from any of the 2 windows machines. The first windows phone 
I start I can send/receve calls the second one I cannot. I. No matter 
which one I start first only the first one works. The linux kphone can 
still call/receive from any of the 2 windows machine. I dont have another 
linux box to see if another kphone could send/receive. Everything seems to 
register fine in asterisks. The 2 windows machines are on seperate servers 
and in the same subnet.  Any ideas?

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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Joe Greco

i am new to this list and i dot not know, if anybody had already the
same problem. I have two cisco 7960 which i want to upgrade to sip. Has
somebody already taken the upgrade-process for special hints and
suggestions? I have already visited the cisco-page and i have read the
proposal for the migration. Is there a special order of firmware-upgrades?

[correct quoting order restored... damn top-posting]

The definitive guide of what versions can be upgraded to what is at:

http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html

In particular, look at tables 2 and 3.

Horrible answer.

Better:

1) Take ANY Cisco documentation with a ton of salt. I've seen numerous
examples of it being broken, silly, and just plain wrong. And that's
just the useful and relevant bits.

2) Run, don't walk, run over to the Wiki and stare at the numerous notes
available on upgrading the firmware on these. Probably a good idea
to look at related pages too.

Some of us have put up information to make it easier for you to get the
dirty work of upgrading one of these phones done. It may not be neat, it
may require some reading and effort, it may require a little trial and
error, but it ought to be all there.

Cisco makes some great phones, but their documentation and their upgrade
processes are crappy.

Don't give up, though, and don't let any consultants talk you in to
paying them to do it for you.

obDisclosure: we do consulting work here. But we believe in end-user
empowerment and we're not interested in upgrading your phones. That's
why I contributed a few missing bits to the Wiki, which definitely can
get you through the whole process, now (I hope)!

Now back to lurking,

... 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.
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work

2005-03-06 Thread Roman Volf

It would be helpful if you pasted the relevant sections of sip.conf and 
extensions.conf

Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]

[EMAIL PROTECTED] wrote:
Im new to Astererisk. I compiled the latest CVS and setup the server. It 
looks like things are working. I'm running kphone, x-lite and sjphone to 
test things out.  The kphone (local to the asterisk server) can call and 
receive calls from any of the 2 windows machines. The first windows phone 
I start I can send/receve calls the second one I cannot. I. No matter 
which one I start first only the first one works. The linux kphone can 
still call/receive from any of the 2 windows machine. I dont have another 
linux box to see if another kphone could send/receive. Everything seems to 
register fine in asterisks. The 2 windows machines are on seperate servers 
and in the same subnet.  Any ideas?

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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread sgup015

Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?

We have just about tried everything that was suggested to us without luck.

Cheers,
Sahil
Quoting Joe Greco [EMAIL PROTECTED]:

[correct quoting order restored... damn top-posting]

The definitive guide of what versions can be upgraded to what is at:

http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html

In particular, look at tables 2 and 3.

Horrible answer.

Better:

1) Take ANY Cisco documentation with a ton of salt. I've seen numerous
examples of it being broken, silly, and just plain wrong. And that's
just the useful and relevant bits.

2) Run, don't walk, run over to the Wiki and stare at the numerous notes
available on upgrading the firmware on these. Probably a good idea
to look at related pages too.

Cisco makes some great phones, but their documentation and their upgrade
processes are crappy.

Don't give up, though, and don't let any consultants talk you in to
paying them to do it for you.

Now back to lurking,

... 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.
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RE: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Nabeel Jafferali

 Whilst I agree with Joe, has anybody actually been able to
 sucessfuly get the Cisco 7940's/7960's to register into *?

Yeah, I've been using a 7960 with * since November.

Nabeel
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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Blake Van Eekeren

Nabeel Jafferali wrote:
Whilst I agree with Joe, has anybody actually been able to
sucessfuly get the Cisco 7940's/7960's to register into *?

Yeah, I've been using a 7960 with * since November.
Indeed, I have 7905/7940/7960's all working with *.
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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Joe Greco

 Hi,
 Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
 Cisco 7940's/7960's to register into *?
 
 We have just about tried everything that was suggested to us without luck.

Um, well, really, that's never been a problem here.  I've had more problems
trying to get them to register directly with VoIP providers than I care to
think about, though.

You need to make sure you've dotted your i's and crossed your t's with
these phones, but then they work really well.

from the SIPmac.cnf:

line2_name:  2002
line2_authname:  2002
line2_password:  khafusulhff
line2_shortname: DisplayedLineName
proxy2_address:  some.ip.addr.ess

sip.conf:

[2002]
type=friend ; This device takes and makes calls
secret=khafusulhff  ; Password for device
auth=md5
host=dynamic; This host is not on the same IP addr every time
username=2002   ; Username programmed into Cisco phone
context=from-7960   ; Inbound calls from this phone go to this context
nat=no  ; nat=yes if this phone is behind a NAT box or firewall
;callgroup=2; the group to which this phone belongs for *8 phone rin
ging pickup
;pickupgroup=2  ; the pickup group allowed from this phone when *8 is di
aled
mailbox=2902; Activate the MW light if this VMB has messages in it


Obviously not a complete configuration.  Note in particular that you 
probably need certain items out of SIPDefaults.

... 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.
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RE: [Asterisk-Users] voicemail volume

2005-03-06 Thread Howard Lowndes

On Mon, 2005-03-07 at 09:02, David Newman wrote:
 On Sun, 6 Mar 2005, Marty Mastera wrote:
 
  The full text of the bug you reference above indicates that pstnVMgain
  was (or is) part of an ongoing feature request/bug report and has not
  been implemented for use at this time (and may never be).
 
 Right. So -- what can I do to boost volume of PSTN - * voicemail?

Assuming you are using a zap interface for the PSTN connection, could
you try increasing the rx gain.  Is your incoming volume low anyway?

 
 thanks
 
 dn
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--
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--
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RE: [Asterisk-Users] Cisco 7960

2005-03-06 Thread sgup015

Any assistance on gettting bi-directional calling going would be great...

We got it working in SIP but it won't register hence calls going to the phones
don't even start..

Quoting Nabeel Jafferali [EMAIL PROTECTED]:

  Whilst I agree with Joe, has anybody actually been able to
  sucessfuly get the Cisco 7940's/7960's to register into *?

 Yeah, I've been using a 7960 with * since November.

 Nabeel
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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
| Hi,
| Whilst I agree with Joe, has anybody actually been able to sucessfuly
get the
| Cisco 7940's/7960's to register into *?
|
| We have just about tried everything that was suggested to us without luck.
|
Yes, working perfectly her with SIP firmware 7.3
- - - - - 8
|Some of us have put up information to make it easier for you to get the
|dirty work of upgrading one of these phones done.  It may not be neat, it
|may require some reading and effort, it may require a little trial and
|error, but it ought to be all there.
|
|Cisco makes some great phones, but their documentation and their upgrade
|processes are crappy.
- - - - 8
I got my phone from eBay with SCCP 3.3 firmware.  To upgrade to SIP 7.3
took about 2 hours of trying to make Cisco's procedure work, before
pouring over the tftp logs then about 1 hour of experimentation and
eventual success.  Now, if I can only remember what I did to get it to
work... ;)
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.6 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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cgOWRgSovq6aDRfKcjzTQ86TDzOvmjEodkuhwCrMpFpH33KZq1wbefR8ZxE4CV0K
Gr+6dYt7WLpGV4QVILfheDnfl1hdNIcaa07kxC2+R+dqsXQ6NU1wv9x5snTE092Y
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w3W8Riu1adS7N/pzUgsR1o9iDNsnWKUJev4kq0WKiF3XToGH5OGFvQ==
=lLW6
-END PGP SIGNATURE-
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RE: [Asterisk-Users] Survey: what's the best HTTPd/TFTPd/FTPd to serveup configuration files to sets

2005-03-06 Thread Jim Van Meggelen

Thanks as always to everyone who provided feedback. It was most helpful!

Regards,

Jim.


--
Jim Van Meggelen
[EMAIL PROTECTED]



[EMAIL PROTECTED] wrote:
 I would like to start a discussion centred around the various
 ways one might serve up configuration files from an Asterisk
 server (I know, it's better to use a secondary server for all
 this, but let's talk about a smaller system).
 
 The types of things being served would include:
 - Logo image for sets that support that
 - XML directory files
 - XML or raw text configuration files
 - what-all-else
 
 Seems to me that Apache is simply way too overpowered for all
 this, and thus would needlessly place load on the server.
 
 I have heard that khttpd is pretty lightweight, but its use
 seems to have been deprecated, and it does not appear to be
 actively maintained. Is TuX the way to go?
 
 As for tftpd and ftpd, I'm just not sure. Leightweight is the
 key, here.
 
 Thoughts, opinions, experiences?
 
 Thanks,
 
 Jim.
 
 
 --
 Jim Van Meggelen
 [EMAIL PROTECTED]

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No virus found in this outgoing message.
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[Asterisk-Users] Music Volume ?

2005-03-06 Thread Mateo Meier

Hey guys

Anybody knows how to turn up the volume of a Music on Hold Mp3 file ?
When I play it on my windows box, volume is perfect.. but when I use it
Music on hold.. the volume is very low.

Maybe there is a general setting for asterisk volume ?

Thx for the help
Matt





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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread C F

I've just intalled a system with 25 Cisco 7960s worked perfectly,
please tell what you did wrong, maybe I can help you.

On Mon, 7 Mar 2005 11:41:38 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?

We have just about tried everything that was suggested to us without luck.

Cheers,
Sahil
Quoting Joe Greco [EMAIL PROTECTED]:

[correct quoting order restored... damn top-posting]

The definitive guide of what versions can be upgraded to what is at:

http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html

In particular, look at tables 2 and 3.

Horrible answer.

Better:

1) Take ANY Cisco documentation with a ton of salt. I've seen numerous
examples of it being broken, silly, and just plain wrong. And that's
just the useful and relevant bits.

2) Run, don't walk, run over to the Wiki and stare at the numerous notes
available on upgrading the firmware on these. Probably a good idea
to look at related pages too.

Cisco makes some great phones, but their documentation and their upgrade
processes are crappy.

Don't give up, though, and don't let any consultants talk you in to
paying them to do it for you.

Now back to lurking,

... 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.
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Re: [Asterisk-Users] Cisco 7960

Works sweet here with a 7960G too, 7.3 SIP fware.
Mike

On Mon, 7 Mar 2005 11:41:38 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Whilst I agree with Joe, has anybody actually been able to sucessfuly get the
Cisco 7940's/7960's to register into *?

We have just about tried everything that was suggested to us without luck.

Cheers,
Sahil
Quoting Joe Greco [EMAIL PROTECTED]:

[correct quoting order restored... damn top-posting]

The definitive guide of what versions can be upgraded to what is at:

http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html

In particular, look at tables 2 and 3.

Horrible answer.

Better:

1) Take ANY Cisco documentation with a ton of salt. I've seen numerous
examples of it being broken, silly, and just plain wrong. And that's
just the useful and relevant bits.

2) Run, don't walk, run over to the Wiki and stare at the numerous notes
available on upgrading the firmware on these. Probably a good idea
to look at related pages too.

Cisco makes some great phones, but their documentation and their upgrade
processes are crappy.

Don't give up, though, and don't let any consultants talk you in to
paying them to do it for you.

Now back to lurking,

... 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.
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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Doug Lytle

[EMAIL PROTECTED] wrote:
Any assistance on gettting bi-directional calling going would be great...
We got it working in SIP but it won't register hence calls going to the phones
don't even start..
 

 

If the phones are behind a NAT, make sure the option on the phone for 
NAT is set to yes.

Doug
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Re: [Asterisk-Users] LiveVoIP Problems?

Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike


On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
 Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
 calls. Calls that are being routed to wrong numbers. DID's that ring
 busy. For the pass 2 days I am unable to pass CID. Is anyone else have
 these problems? Can anyone recommend a Quality VoIP provider?
 
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[Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)

2005-03-06 Thread Maxim Litnitsky

Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config

if (uri==myself) {
if (method==REGISTER) {
save(location);
log (1, Registered\n);
break;
};
if (lookup(location)) {
 log (1, ***  IP to IP call *);
 if (method == INVITE){
 setflag (1);
 t_on_failure(1);
 t_relay();
 sl_send_reply (180, Ringing);
setflag (1);
 break;
 }
 if (!t_relay()) {
  sl_send_reply(404, Not Found);
  break;
 };

#};
break;
};


failure_route[1] {
revert_uri();
forward(69.70.x.x,5060);
break();
}

Asterisk sip.conf:

[ser]
host=69.70.x.x
context=ser
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=gsm
allow=ilbc
nat=yes

extensions.conf:

[ser]
include = vm
include = messagecenter

[vm]
exten = _9.,1,VoiceMail(u${EXTEN})
exten = _9.,2,Hangup

[messagecenter]
exten = 555,1,Answer
exten = 555,2,Wait(1)
exten = 555,3,VoiceMailMain(default)
exten = 555,4,Hangup
exten = _555X.,1,Answer; can dial 555exten
to skip 'mailbox' prompt.  Useful for speedial.
exten = _555X.,2,Wait(1)
exten = _555X.,3,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
exten = _555X.,4,Hangup


All SER calls  9xxx must go to asterisk, and it does, but I get the
following in aster log:
 to 69.70.7.174:5060
Mar  6 18:41:36 WARNING[3539]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for seqno 1
(Non-critical Response)
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav49, 0x814cb60
-- x=1, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: gsm, 0x814d068
-- x=2, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav, 0x8144980
Mar  6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio
available on SIP/69.70.x.x-08149a98??
-- User hung up
  == Spawn extension (ser, 900, 1) exited non-zero on 'SIP/69.70.x.x-08149a98'
Destroying call '[EMAIL PROTECTED]'


If I use rewritehostport instead of forward, the call does not reach asterisk:

failure_route[1] {
revert_uri();
rewritehostport(69.70.x.x:5060);
t_relay()
break();

SER log:

4(11513) ***  IP to IP call * 1(11506) ERROR:
t_forward_nonack: no branched for fwding
 1(11506) ERROR: w_t_relay (failure mode): forwarding failed
 3(11512) ***  IP to IP call * 2(11509) Bye

Is there a way to do append_branch([EMAIL PROTECTED]) ?


Anyone did it? Reply pls with your config files!!
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work

the kphone is using 214 and the windows 204 and 203. It doesnt matter 
though I can have kphone use 203 and windows 214,204 and get the same 
issues.

sip.conf:
[214]
type=friend
username=214
secret=214
callerid=test 214
nat=no
canreinvite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mailbox=204
host=dynamic
 
[203]
type=friend
username=203
secret=203
callerid=test 203
nat=no
canreinvite=yes
disallow=all
allow=alaw
allow=gsm
allow=ulaw
mailbox=203
host=dynamic
 
[204]
type=friend
username=204
secret=204
callerid=test 204
nat=no
canreinvite=yes
disallow=all
allow=gsm

extensions.conf:
exten = 204,1,Dial(SIP/204,20,rt)
exten = 204,n,Voicemail(u204)
exten = 204,s+1,Hangup
exten = 214,1,Dial(SIP/214,20,rt)
exten = 214,n,Voicemail(u214)
exten = 214,s+1,Hangup
exten = 203,1,Dial(SIP/203,20,rt)
exten = 203,n,Voicemail(u203)
exten = 203,s+1,Hangup





Roman Volf [EMAIL PROTECTED] 
Sent by: [EMAIL PROTECTED]
03/06/2005 05:38 PM
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com


To
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asterisk-users@lists.digium.com
cc

Subject
Re: [Asterisk-Users] Trying to get 2 SIP phones to work






It would be helpful if you pasted the relevant sections of sip.conf and 
extensions.conf

Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]



[EMAIL PROTECTED] wrote:

Im new to Astererisk. I compiled the latest CVS and setup the server. It 
looks like things are working. I'm running kphone, x-lite and sjphone to 
test things out.  The kphone (local to the asterisk server) can call and 
receive calls from any of the 2 windows machines. The first windows phone 

I start I can send/receve calls the second one I cannot. I. No matter 
which one I start first only the first one works. The linux kphone can 
still call/receive from any of the 2 windows machine. I dont have another 

linux box to see if another kphone could send/receive. Everything seems 
to 
register fine in asterisks. The 2 windows machines are on seperate 
servers 
and in the same subnet.  Any ideas?

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Re: [Asterisk-Users] Cisco 7960

2005-03-06 Thread Alistair Cunningham

Joe Greco wrote:
The definitive guide of what versions can be upgraded to what is at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guides09186a008022a968.html
In particular, look at tables 2 and 3.

Horrible answer.
Better:
1) Take ANY Cisco documentation with a ton of salt. I've seen numerous
examples of it being broken, silly, and just plain wrong. And that's
just the useful and relevant bits.
2) Run, don't walk, run over to the Wiki and stare at the numerous notes
available on upgrading the firmware on these. Probably a good idea
to look at related pages too.
I assure you that this page is what the original poster wants, at least
to begin with. I've used it to upgrade quite a few 7960s. I've never had
one that didn't work, though I'll admit that some did take a couple of
tries or intermediary steps. Follow table 3 on that page, and if it
doesn't work, then try the Wiki.

I would agree that information is hard to find on Cisco's website, and
it's often contradictory, but in this case that is the page to follow.

We do consulting work as well, in fact it's our main business, but
frankly for 2 phones the rate we'd charge would not be economical. If
you had 200 or 2000 then we could give you a sensible quote.

In answer to another poster, 7960s running SIP can register with
Asterisk with no problems.

--
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
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Re: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread Eric Wieling aka ManxPower

Mike Dent wrote:
Makes you wonder how many *really* reliable VoIP providers there are out there?
Who would you trust to handle all your incoming/outgoing business calls?
None.
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Re: [Asterisk-Users] SER - Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)

2005-03-06 Thread Andres



If I use rewritehostport instead of forward, the call does not reach asterisk:
failure_route[1] {
   revert_uri();
   rewritehostport(69.70.x.x:5060);
   t_relay()
   break();
SER log:
 

Your failure route should read:
failure_route[1] {
   revert_uri();
   rewritehostport(69.70.x.x:5060);
   append_branch();   ==YOU MISSED THIS 
   t_relay()
   break();


--
Andres
Network Admin
http://www.telesip.net
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[Asterisk-Users] IP Providers pass CallerID?

2005-03-06 Thread TELUX

Are there any IP Providers that will pass Caller ID? Broadvoice used to 
but no they dont.

THX  
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work

2005-03-06 Thread Time Bandit

 receive calls from any of the 2 windows machines. The first windows phone
 I start I can send/receve calls the second one I cannot. I. No matter
 which one I start first only the first one works. The linux kphone can

Please take note that each phone need it's account. You can't have 2
phone registering with the same account.

hth
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[Asterisk-Users] Re: [Asterisk-biz] Livevoip U.S. 800 LNP Starts March 9th 2005

Mike,

No they have not. Calls are failing again today. They have offered to
refund my money but that does not solve the problem. My asterisk server
is only 4 to 12 ms away from their network. I have had VERY good luck
with nufone.(40 to 45ms away) Only have 1 or 2% fail rate. Going to be
calling txlink.net on Monday. 

Seems that LiveVoIP does not care about asterisk users. They like to
pass the blame. 

-Tim

On Sun, 2005-03-06 at 17:04, Mike Dent wrote:
 Hmmm, I was contemplating going with livevoip, glad I read your post.
 I'd be interested if they resolved your issues?
 
 thanks
 Mike
 
 
 
 On Fri, 04 Mar 2005 22:45:58 -0600, Tim [EMAIL PROTECTED] wrote:
  Instead of offering new services. Why don't you get the ones that you do
  offer to work right first!
  
  Outstanding Problems
  
  1. IAX ring back
  
  2. DID's that don't work half the time
  
  3. Caller ID
  
  4. missing DTMF 50% of the time.
  
  5. outgoing call that are routed to the wrong numbers.
  
  
  On Fri, 2005-03-04 at 22:21, Brandon Patterson wrote:
   LiveVoip LLC will offer 800 LNP Number Porting Starting March 9th.
  
   800 Number Porting information will be posted in detail on our site
   late Sat evening March 4, 2005. People have been asking for this
   service
   and we are happy to provide it. This is made possible by various CLEC
   relationships, upgraded switch capacity, and extra staffing. This
   service
   is for U.S. 800 numbers only.
  
   LiveVoip LLC
   Connect Locally - Talk Globally
  
   http://www.livevoip.com
   [EMAIL PROTECTED][EMAIL PROTECTED]
  
   800 Team Email: [EMAIL PROTECTED] or [EMAIL PROTECTED]
  
   __
  
  ___
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  http://lists.digium.com/mailman/listinfo/asterisk-biz
 

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[Asterisk-Users] Cisco 7960

2005-03-06 Thread Peter Illmayer

 Date: Sun, 06 Mar 2005 20:03:52 +0100
 From: Thomas Trepper [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Cisco 7960
 To: asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=us-ascii; format=flowed

 Hi all,

 i am new to this list and i dot not know, if anybody had already the 
 same problem. I have two cisco 7960 which i want to upgrade to sip. 
 Has somebody already taken the upgrade-process for special hints and 
 suggestions? I have already visited the cisco-page and i have read 
 the proposal for the migration. Is there a special order of firmware-
 upgrades?

 Thanks a lot

 Thomas

Thomas

The asterisk-wiki is the best place to start.  It will tell you that it is a 3
stage process if your currently on call-manager.

You will need to load the version 3, then 5 and then 7 SIP firmware.  I tried
to load the version 7 straight away and of course it wouldnt work.

Please read the wiki and all will be revealed.  Dont expect very much from the
cisco website at all !

Regards..pete
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[Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing

2005-03-06 Thread Cameron Beattie





I am based in New Zealand and 
am experiencing the same problem as referred to in the post "FXO module in 
TDM400P (UK, BT) - Hangup detection failing" from 2 November 2004 i.e. Zap/4 
(being the FXO module) not detecting hangup on the PSTN line if the call is not 
answered on a PABX extension. 

Has anyone managed to find a 
resolution to the problem?

For information:
Digium TDM400P with FXS on Zap 1 
2 and FXO on Zap 3  4.
CVS-v1-0-01/24/05
Using fxs_ks signalling

Regards

Cameron
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[Asterisk-Users] RE: Need help on * anf HFC.

2005-03-06 Thread MvB

Hi,

See comments in line

 configured at mem 0xf91c5e00 fifo 0xf7598000(0x37598000) IRQ 5 HZ 100
 Mar  6 21:29:13 linux-1 kernel: zaphfc: Card 0 configured for NT mode
 Mar  6 21:29:13 linux-1 kernel: zaphfc: 1 hfc-pci card(s) in this box.

Your card is configured in NT mode this something you do when you
connect to a TE device. When you connect to the ISDN BRI from your telco
(i.e. connect your NT1 telco box to your asterisk) then the card needs
to talk in TE mode. (which will act in TE mode)

 ; p2p NT mode
 signalling = bri_net
 pridialplan = local

I think this should be changed to signalling = bri_cpe_ptmp. As well it
may be that your pridialplan needs to be changed, this is different per
telco and/or region. You can check this when you dial out and you have
your bri debugging on. If it is wrong then dialing out will fail ;-)

You also may want to check your /etc/zaptel.conf as it is now
registering tones for the Netherlands.

Hope this helps.

Max.




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Re: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread The Phone Guys

So you want it 100% perfect and you want it for peanuts.
Makes you wonder how many *really* reliable VoIP providers there are out 
there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike

On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
calls. Calls that are being routed to wrong numbers. DID's that ring
busy. For the pass 2 days I am unable to pass CID. Is anyone else have
these problems? Can anyone recommend a Quality VoIP provider?
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Re: [Asterisk-Users] Cisco 7960

On Mon, 7 Mar 2005, Peter Illmayer wrote:
You will need to load the version 3, then 5 and then 7 SIP firmware.  I tried
to load the version 7 straight away and of course it wouldnt work.
FWIW, I have also had success doing versions 3, 6, and then 7 in moving 
from Skinny to SIP. But it's still three steps.

dn
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Re: [Asterisk-Users] LiveVoIP Problems?

No. When DID go down for a whole day. Do you think thats okay?  Ring
busy half time or do nothing at all. Come on! Your DID's are up maybe
50% of the time if that! 


 Why are calls failing again today?



On Sun, 2005-03-06 at 17:36, The Phone Guys wrote:
 So you want it 100% perfect and you want it for peanuts.
 
  Makes you wonder how many *really* reliable VoIP providers there are out 
  there?
  Who would you trust to handle all your incoming/outgoing business calls?
  Mike
 
 
  On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
  Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
  calls. Calls that are being routed to wrong numbers. DID's that ring
  busy. For the pass 2 days I am unable to pass CID. Is anyone else have
  these problems? Can anyone recommend a Quality VoIP provider?
 
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Re: [Asterisk-Users] LiveVoIP Problems?

No, I dont mind paying more for something if I know its going to be reliable.


On Sun, 6 Mar 2005 16:36:16 -0700, The Phone Guys [EMAIL PROTECTED] wrote:
 So you want it 100% perfect and you want it for peanuts.
 
  Makes you wonder how many *really* reliable VoIP providers there are out
  there?
  Who would you trust to handle all your incoming/outgoing business calls?
  Mike
 
 
  On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
  Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
  calls. Calls that are being routed to wrong numbers. DID's that ring
  busy. For the pass 2 days I am unable to pass CID. Is anyone else have
  these problems? Can anyone recommend a Quality VoIP provider?
 

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RE: [Asterisk-Users] voicemail volume


On Mon, 7 Mar 2005, Howard Lowndes wrote:
On Mon, 2005-03-07 at 09:02, David Newman wrote:
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that pstnVMgain
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN - * voicemail?
Assuming you are using a zap interface for the PSTN connection, could
you try increasing the rx gain.  Is your incoming volume low anyway?
Generally seems OK, but I'm not an audio engineer and I don't know how to 
take measurements to quantify low. The problem is MUCH more perceptible 
on voicemail, esp. on the .wav email attachments.

dn
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Re: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread John Novack


The Phone Guys wrote:
So you want it 100% perfect and you want it for peanuts.
OF COURSE!
They all certainly imply and promise that.
Would anyone subscribe if they said  we have a second rate service  ?

Makes you wonder how many *really* reliable VoIP providers there are 
out there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike

On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed 
outgoing
calls. Calls that are being routed to wrong numbers. DID's that ring
busy. For the pass 2 days I am unable to pass CID. Is anyone else have
these problems? Can anyone recommend a Quality VoIP provider?

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RE: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread Roman Zhovtulya

What do folks have to say about www.voipjet.com?
(IAX, call termination only)



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Montag, 7. März 2005 00:58
To: The Phone Guys; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] LiveVoIP Problems?




The Phone Guys wrote:

 So you want it 100% perfect and you want it for peanuts.

OF COURSE!
They all certainly imply and promise that.
Would anyone subscribe if they said  we have a second rate service  ?



 Makes you wonder how many *really* reliable VoIP providers there are
 out there?
 Who would you trust to handle all your incoming/outgoing business
calls?
 Mike


 On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:

 Anyone having problems with LiveVoIP lately? I am seeing failed
 outgoing
 calls. Calls that are being routed to wrong numbers. DID's that ring
 busy. For the pass 2 days I am unable to pass CID. Is anyone else
have
 these problems? Can anyone recommend a Quality VoIP provider?


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Re: [Asterisk-Users] LiveVoIP Problems?

Looks like if you complain. They terminate your account. Or is this just
another BUG

Mar  6 17:57:56 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar  6 17:57:56 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher' rejected: Registration Refused
Mar  6 17:58:46 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar  6 17:58:46 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher' rejected: Registration Refused
Mar  6 17:59:36 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar  6 17:59:36 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher' rejected: Registration Refused
Mar  6 18:00:26 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar  6 18:00:26 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher' rejected: Registration Refused
Mar  6 18:01:16 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'wdsmn' rejected: Registration Refused
Mar  6 18:01:16 NOTICE[18108]: chan_iax2.c:6695 socket_read:
Registration of 'tschacher' rejected: Registration Refused


On Sun, 2005-03-06 at 17:57, John Novack wrote:
 The Phone Guys wrote:
 
  So you want it 100% perfect and you want it for peanuts.
 
 OF COURSE!
 They all certainly imply and promise that.
 Would anyone subscribe if they said  we have a second rate service  ?
 
 
 
  Makes you wonder how many *really* reliable VoIP providers there are 
  out there?
  Who would you trust to handle all your incoming/outgoing business calls?
  Mike
 
 
  On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
 
  Anyone having problems with LiveVoIP lately? I am seeing failed 
  outgoing
  calls. Calls that are being routed to wrong numbers. DID's that ring
  busy. For the pass 2 days I am unable to pass CID. Is anyone else have
  these problems? Can anyone recommend a Quality VoIP provider?
 
 
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Re: [Asterisk-Users] LiveVoIP Problems?

Maybe at some stage in the future the big telcos will provide VoIP
termination, DID's etc. They may as well make some money from it, I'm
sure they could get it right?

BT providing IAX2 and SIP termination? Hmmm, maybe one day.

Mike
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[Fwd: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound]

2005-03-06 Thread MF Hulber







 Original Message 

  

  Subject: 
  Re: [Asterisk-Users] BroadVoice configuration changes for
Outbound


  Date: 
  Sun, 06 Mar 2005 19:11:22 -0500


  From: 
  MF Hulber [EMAIL PROTECTED]


  To: 
  Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, [EMAIL PROTECTED]


  References: 
  [EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]

  



Well, I tried these instructions and still get the following on outgoing:

Mar  6 19:08:16 NOTICE[-1291998288]: chan_sip.c:5047 handle_response: 
Failed to authenticate on INVITE to '"sipura1_1" 
sip:[EMAIL PROTECTED];tag=XX'


Dan Weber wrote:

 In the last email I sent, I did not mean to insult anyone, but I have 
 tested the instructions thoroughly I provided.  If you were using the 
 instructions I provided originally, you would not be able to make 
 outbound calls.  Here are the instructions that have been known to work;
 Please read line by line and setup that way.

 http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup

 Dan

 On Sun, 6 Mar 2005, skamp wrote:

 sorry still doesnt help with incoming calls, there is definatley
 something more wrong, my config was working fine until today and its
 worked fine for months. They have broken something.

 On Sun, 2005-03-06 at 02:23 -0600, Bartosz Wegrzyn - asterisk wrote:

 [broadvoice-incoming]
 type=peer
 host=147.135.8.128
 context=from-broadvoice
 qualify=yes
 canreinvite=no
 disallow=all
 allow=ulaw
 nat=never

 [broadvoice-incoming2]
 type=peer
 host=147.135.0.128
 context=from-broadvoice
 qualify=yes
 canreinvite=no
 disallow=all
 allow=ulaw
 nat=never

 [broadvoice-incoming3]
 type=peer
 host=147.135.4.128
 context=from-broadvoice
 qualify=yes
 canreinvite=no
 disallow=all
 allow=ulaw
 nat=never

 -- 
 skamp [EMAIL PROTECTED]

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Re: [Asterisk-Users] SpanDSP: Training failed (sequence failed)

2005-03-06 Thread Steve Underwood

CClarke wrote:
Hello All ~
Having problems sending and receiving faxes with SpanDSP. I am testing on a
simple 2 analog POTS to 2x X100p set up, connecting one line to a Konica 720
fax machine to test, or with other remote fax machines. Voice calls are
working pretty well now. Platform is P3/800MHz/256MB/FC1.
* recognizes faxes, and passes calls to RxFax and TxFax OK, but 99.9% of the
time no fax content is sent or received. (On one occasion coincidentally a
 

So, you tried at least 1000 tests to get this statistic. Very dedicated. :
junk fax came in successfully while I was testing!).
I've reviewed previous posts and but can't find any relevant advice on where
to go next, since the fax negotiation seems to go OK so far, but then die (see
below), and no content is transmitted. The final error message from RxFax is:
Training failed (sequence failed)
Would really appreciate some expert advice on what this means and how to
fix...
Christina.
PS: Previous somewhat relevant posts I've been able to find:
http://lists.digium.com/pipermail/asterisk-users/2004-June/051143.html
http://lists.digium.com/pipermail/asterisk-users/2005-February/090978.html
 

Those don't seem very relevant. There are plenty of posts which are.
Here's versions I'm using:
asterisk CVS-v1-0-02/20/05-17:04:48
spandsp 0.0.1k
libtiff 3.5.7
 

Try using spandsp-0.0.2pre10. spandsp-0.0.1k is now very old. That 
shouldn't be very relevant to your problem, since the modems in 0.0.1k 
usually decode reliably. The ones in 0.0.2pre10 are more tolerant of 
really bad phone lines, though. Also, I have no intention of supporting 
0.0.1k any more.

The fast modem isn't training properly. A number of people report this, 
and there is nothing I can do about it. They have problems in their * 
setup, which prevents spandsp from getting a clean signal. I really need 
to put a self-diagnosis feature in spandsp so it can detect and report 
these problems.

Regards,
Steve
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work


each phone logs in under its own sip
account: 203, 204 and 214. I assume the account is whats in the sip.conf
file.





Time Bandit [EMAIL PROTECTED]

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03/06/2005 06:22 PM



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SIP phones to work








 receive calls from any of the 2 windows machines.
The first windows phone
 I start I can send/receve calls the second one I cannot. I. No matter
 which one I start first only the first one works. The linux kphone
can

Please take note that each phone need it's account. You can't have 2
phone registering with the same account.

hth
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[Asterisk-Users] Re: Broadvoice configuration changes for outbound calls

2005-03-06 Thread Brian Buhrow

Hello.  I'm not sure what's going on with the gentleman who is having
trouble receiving inbound calls as of this weekend, but I can say that
while inbound works for me, calling out through BroadVoice doesn't work at
all.  SIP traces show that when I send an invite request out to BroadVoice,
they send back a 401 unauthorized message which includes a
WWW-Authentication: header which ASterisk is supposed to use to send a
reply proxy authentication response.
The version of Asterisk I'm running, and have been running with
BroadVoice for months claims that it sends an acknowledgement of the
unauthorized message, then fails to send an authentication reply, instead
claiming that authentication is impossible with BroadVoice.
I suspect that there is a bug in the md5 hashing  code on the version
of Asterisk I'm running, and I'll be attempting to upgrade things, or sort
out the bug soon.
My point here is to let people know that they may be seeing different
behaviors depending on what version of ASterisk code they're running.  I'm
running with CVS head as of 2003-12-18.  I doubt many others are running
code this old, but until this Saturday morning, it's worked flawlessly with
every provider I've tried it with.
Having said all that, I too am disappointed that BroadVoice has not
seen fit to tell its users of this impending change.  Instead, it worked on
Friday night for me, all normal, and, voila! complete failure of outgoing
calls on Saturday morning.  Most disturbing.

Hope that's somewhat helpful.
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Re: [Asterisk-Users] Music Volume ?

2005-03-06 Thread C F

Check out musiconhold.conf
you can use loud


On Mon, 7 Mar 2005 00:02:27 +0100, Mateo Meier [EMAIL PROTECTED] wrote:
 Hey guys
 
 Anybody knows how to turn up the volume of a Music on Hold Mp3 file ?
 When I play it on my windows box, volume is perfect.. but when I use it
 Music on hold.. the volume is very low.
 
 Maybe there is a general setting for asterisk volume ?
 
 Thx for the help
 Matt
 
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work

All fixed. I just updated from CVS, rebuild and everything works. I did 
try restarting astrisks before I tried this so it either didnt pick up a 
config right or the new CVS fixed it.




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Asterisk Users Mailing List - Non-Commercial Discussion 
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Subject
Re: [Asterisk-Users] Trying to get 2 SIP phones to work







each phone logs in under its own sip account: 203, 204 and 214. I assume 
the account is whats in the sip.conf file. 


Time Bandit [EMAIL PROTECTED] 
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Re: [Asterisk-Users] Trying to get 2 SIP phones to work








 receive calls from any of the 2 windows machines. The first windows 
phone
 I start I can send/receve calls the second one I cannot. I. No matter
 which one I start first only the first one works. The linux kphone can

Please take note that each phone need it's account. You can't have 2
phone registering with the same account.

hth
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Re: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread Paul Fielding

*shrug*.  Mine's been working flawlessly since I've had it (~month).  The 
only 2 issues I have are the ringback problem, and I can only send callerid 
number info to them, not name info  Guess we'll see how long it 
lasts

regards,
Paul
- Original Message - 
From: Tim [EMAIL PROTECTED]
To: The Phone Guys [EMAIL PROTECTED]; Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Sunday, March 06, 2005 4:45 PM
Subject: Re: [Asterisk-Users] LiveVoIP Problems?

No. When DID go down for a whole day. Do you think thats okay?  Ring
busy half time or do nothing at all. Come on! Your DID's are up maybe
50% of the time if that!
Why are calls failing again today?

On Sun, 2005-03-06 at 17:36, The Phone Guys wrote:
So you want it 100% perfect and you want it for peanuts.
 Makes you wonder how many *really* reliable VoIP providers there are 
 out
 there?
 Who would you trust to handle all your incoming/outgoing business 
 calls?
 Mike

 On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
 Anyone having problems with LiveVoIP lately? I am seeing failed 
 outgoing
 calls. Calls that are being routed to wrong numbers. DID's that ring
 busy. For the pass 2 days I am unable to pass CID. Is anyone else have
 these problems? Can anyone recommend a Quality VoIP provider?

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Re: [Asterisk-Users] LiveVoIP Problems?

2005-03-06 Thread Paul

No, I want it to work 50% of the time and pay half your current pricing. 
Or maybe we can make this really easy for you to understand. Make it 
work 0% of the time and we pay you nothing.

I think that people expect it to work about 99.99% of the time if they 
are going to use it for production purposes. Get your act together and 
raise prices if necessary in order to keep it together.

The Phone Guys wrote:
So you want it 100% perfect and you want it for peanuts.
Makes you wonder how many *really* reliable VoIP providers there are 
out there?
Who would you trust to handle all your incoming/outgoing business calls?
Mike

On Fri, 04 Mar 2005 21:18:41 -0600, Tim [EMAIL PROTECTED] wrote:
Anyone having problems with LiveVoIP lately? I am seeing failed 
outgoing
calls. Calls that are being routed to wrong numbers. DID's that ring
busy. For the pass 2 days I am unable to pass CID. Is anyone else have
these problems? Can anyone recommend a Quality VoIP provider?

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RE: [Asterisk-Users] IP Providers pass CallerID?

2005-03-06 Thread James Pooton

That's surprising; I thought they were one of the larger outfits.  I have
tried quite a few for outbound lately and the only one that has reliably
passed Caller ID (using our 406 area code) is simpletelecom.com.  They are
also the only ones I've tried that respond to a support ticket in a
reasonable amount of time.

What is really odd is that I can pass caller ID with iax.cc/Sixtel in other
area codes, but nothing in 406 which I need to do. (Our local numbers)
Anyone know why that would be ?

-James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TELUX
Sent: Sunday, March 06, 2005 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IP Providers pass CallerID?

Are there any IP Providers that will pass Caller ID? Broadvoice used to 
but no they dont.

THX  
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Re: [Asterisk-Users] LiveVoIP Problems?