[Asterisk-Users] Can Asterisk act like a normal sip phone?

[Birk Bremer]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everyone,

I'm relatively new to the subject - so pleace don't punish me for
idiotic questions. ;-)
Can Asterisk act like a normal Sip phone and e.g. connect to another
sip-gateway?  Background: There is a new german company at:
http://www.sipgate.de  (sorry German only page)
They offer a a gateway between a real telephone number and their sip
server. (at the moment for free) If you had the possibility to connect
asterisk as a phone to this server it would be an easy (and cheap!) way
to realise a gateway to old-style-phoneline.
Waiting for reply,

		Birk Bremer



-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAGg6+7QhrwFQeHVsRAo+1AJ9+gk79nIxbxt6rPPpHIBw2MZibBQCdEcJN
wWawRjIjmpUs9orqrmEEcNI=
=EGhT
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Kannaiyan Natesan]

You can use,

;sip.conf

register = username:[EMAIL PROTECTED]/extension

to make asterisk as a SIP client.

to forward calls to another client use  canreinvite=yes, (if the client
supports reinvite)

and in the extensions.conf

exten = s,1,Dial(SIP/username:[EMAIL PROTECTED])

Kannaiyan


- Original Message -
From: Birk Bremer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 7:58 AM
Subject: [Asterisk-Users] Can Asterisk act like a normal sip phone?


 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Hello everyone,

 I'm relatively new to the subject - so pleace don't punish me for
 idiotic questions. ;-)

 Can Asterisk act like a normal Sip phone and e.g. connect to another
 sip-gateway?  Background: There is a new german company at:
 http://www.sipgate.de  (sorry German only page)
 They offer a a gateway between a real telephone number and their sip
 server. (at the moment for free) If you had the possibility to connect
 asterisk as a phone to this server it would be an easy (and cheap!) way
 to realise a gateway to old-style-phoneline.


 Waiting for reply,

 Birk Bremer



 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.2.3 (GNU/Linux)
 Comment: Using GnuPG with Debian - http://enigmail.mozdev.org

 iD8DBQFAGg6+7QhrwFQeHVsRAo+1AJ9+gk79nIxbxt6rPPpHIBw2MZibBQCdEcJN
 wWawRjIjmpUs9orqrmEEcNI=
 =EGhT
 -END PGP SIGNATURE-

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Re: [Asterisk-Users] Introducing Firefly

[Andy Powell]

Hi,

I downloaded this the other day and finally got it to stop crashing. It appears that 
any response from asterisk
that implies an error (for example dialing a non-existant number, using the wrong 
password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash. I've only tested 
the IAX proto not sip or your
own network. running XP with uptodate patches on a local lan.

When it works it works really well, although I don;t particularly like in initial beep 
and end beep when i make
a call (I haven't played with all the options so it may be that I can turn this off).. 
sound quality is good. All in all
a nice little app. Are you planning on allowing other people to run your server side 
(like Jabber does) in their
environments?

If you need any further debugging info on the crashes, let me know...

HTH

Andy


*** REPLY SEPARATOR  ***

On 28/01/2004 at 12:11 Adam Hart wrote:

After many months of development, I'm pleased to announced Firefly - an
IAX soft phone and network.

The firefly softphone - free, runs under windows, allows connection to
multiple networks, skinable interface, connection to firefly network, IAX2
protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw,
GSM. - contact lists, selectable ringtones.

download from here - http://www.virbiage.com/firefly/

The firefly network - also free, runs on an enhanced version of IAX2
(simply uses IAX2 text messages for customised part), voicemail, text
messaging, online presence, ability to indicate status (available, away,
NA). I believe you can connect using a standard asterisk box but you'll
miss out on the extended features. The network runs on iLBC so
unforunately it won't work with most IAX2 clients (unless you get * to
translate)

Thousands of people have used it but it's still regarded in beta, as we
are still in heavy development (so expect a few bugs). It doesn't use
iaxcomm as we needed our own framework to support sip, including our own
jitterbuffer. If you don't wish to connect to the firefly network, click
cancel when it asks you.

Coming soon features
SIP - in alpha, few bugs outstanding
music onhold - playing mp3s while the other party is onhold
fast audio - will reduce the latency by 40-50ms
speex - (if anyone wants it?)

Feel free to contact me on or off the list to report bugs and suggestions.
I'll post everytime we release a new version (probably every week),
including fixed bugs and new features

Our website is http://www.virbiage.com/


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Re: [Asterisk-Users] asterisk with big number of extentions.

[Fran Boon]

Anton wrote:
you can do it with a well setup cluster
OK, so what success have people had with which clustering technologies?

I'm more interested in resilience than performance.

Thanks a lot,
F
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Re: [Asterisk-Users] Expire old voice mail messages, et al

[Philipp von Klitzing]

Hi!

 Also, does anyone feel a need to have the voicemail system speak the
 date and time the voice mail message arrived for those that access
 messages by phone instead of the usual email? 

Did you look at voicemail.conf and the tz= settings? Simply create a 
timezone that fits your needs.

 Finally...am I the only person who does not have a need for separate busy 
 and no answer outgoing messages?

That's especially true when a non-registered SIP client is reported as 
on the phone... anyway you have full control over this in your dialplan 
through Voicemail(1000) = no announcement or Voicemail(b1000) or 
Voicemail(u1000).

Cheers, Philipp


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Re: [Asterisk-Users] Introducing Firefly

[FastJack]

Hi,

just installed Firefly. Looks great, sound is also great. I just got the
following problem.
I'm using Firefly with my asterisk*-box. When I enter a contact with the
number +00233612345 Firefly just erases the 00 when I restart it. Am I
missing something?

Thanks! Great software!!!

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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Walter Doerr]

On Fri, Jan 30, 2004 at 08:20:48AM -, Kannaiyan Natesan wrote:
 You can use,
 
 ;sip.conf
 
 register = username:[EMAIL PROTECTED]/extension
 
 to make asterisk as a SIP client.
 
[...]
 
  Can Asterisk act like a normal Sip phone and e.g. connect to another
  sip-gateway?  Background: There is a new german company at:
  http://www.sipgate.de  (sorry German only page)

I set up an account with sipgate yesterday evening and tried to use the above mentioned
register in sip.conf * to login to sipgate.
No luck so far.

They use SER and I get 483 too many hops replies back from them.

Any help is greatly appreciated.

-Walter



-- 
  Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
  The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Kannaiyan Natesan]

I think there is a loopback.
Did you debug that with sip debug in console and look at SIP Messages what
is doing ?

Kannaiyan


- Original Message -
From: Walter Doerr [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:42 AM
Subject: Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?


 On Fri, Jan 30, 2004 at 08:20:48AM -, Kannaiyan Natesan wrote:
  You can use,
 
  ;sip.conf
 
  register = username:[EMAIL PROTECTED]/extension
 
  to make asterisk as a SIP client.
 
 [...]
  
   Can Asterisk act like a normal Sip phone and e.g. connect to another
   sip-gateway?  Background: There is a new german company at:
   http://www.sipgate.de  (sorry German only page)

 I set up an account with sipgate yesterday evening and tried to use the
above mentioned
 register in sip.conf * to login to sipgate.
 No luck so far.

 They use SER and I get 483 too many hops replies back from them.

 Any help is greatly appreciated.

 -Walter



 --
   Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
   The poor folks who only have 100MBytes of RAM five years
 from now may not be able to buffer a 16MB packet, but that's their
 tough luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Walter Doerr]

On Fri, Jan 30, 2004 at 09:51:44AM -, Kannaiyan Natesan wrote:
 I think there is a loopback.

Or, their SER forwards packets to itself. Hard to tell without knowing
their config.

 Did you debug that with sip debug in console and look at SIP Messages what
 is doing ?

Yes.


-Walter



-- 
  Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
  The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Klaus-Peter Junghanns]

hi,

just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)

make sure you have in sip.conf:

register = :[EMAIL PROTECTED]/extension in your context

you will get the too many hops if you try to register
with their proxy (proxy.de.sipgate.net).


best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/



 I set up an account with sipgate yesterday evening and tried to use the
 above mentioned register in sip.conf * to login to sipgate.
 No luck so far.

 They use SER and I get 483 too many hops replies back from them.

 Any help is greatly appreciated.

 -Walter



 --
   Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
   The poor folks who only have 100MBytes of RAM five years
 from now may not be able to buffer a 16MB packet, but that's their tough
 luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] Re: Asterisk and gnugk (bam)

[bam]

The phone works fine with oh323, its just the need to authenticate the 
endpoint and match a non-fixed ip to a number that has sent me off in the 
direction of gnugk. If I could do it all in * I would.

thanks,

brian

At 18:05 29/01/04, Roger wrote:
Hi,

I also had some problems using chan_oh323 together
with gnugk.
* - gnugk - h323-phone
When I called the phone and hang up, befor the phone
was picked up, the h323-phone continued ringing.
The same, when the h323- and some sip-phones were
called, and the sip-phone picked up the call first.
(It is annoying, when you are talking to someone at
the phone and the phone on the neighbour desk does not
stop ringing!)
Now, I switched to chan_h323 and the h323-phone
works better.
The only problem what remained, is that the phone and
* sometimes don't manage to negotiate a codec both
are supporting. But when gnugk is not in routed
mode, everything is fine!
Roger.

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[Asterisk-Users] Music on Hold Warnings

[Craig Waddington]

Hi.



I am having the following warning when using music on hold.



It works from X-Lite to Grandstream. I get a lot of errors
and warnings.



1.Warning, flexibel rate not heavily tested!



2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread:
Request to schedule in the past?!?!



Thanks for any help.





Full Output below:



Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 909 (Response)

Jan 30 10:24:55 WARNING[1217602880]: file.c:521 ast_readaudio_callback:
Failed to write frame

 == Spawn extension (sip, 5001, 2) exited non-zero on
'SIP/5002-0922'

 -- SIP/5001-6a4d answered SIP/5002-d365

 -- Attempting native bridge of SIP/5002-d365 and
SIP/5001-6a4d

 -- Started music on hold, class 'default', on
SIP/5001-6a4d

Warning, flexibel rate not heavily tested!

Jan 30 10:25:14 NOTICE[1100258240]: res_musiconhold.c:260
monmp3thread: Request to schedule in the past?!?!

 -- Stopped music on hold on SIP/5001-6a4d

 == Spawn extension (sip, 5001, 1) exited non-zero on
'SIP/5002-d365'



 -- Executing Dial(SIP/5002-a28b,
SIP/5001|20) in new stack

 -- Called 5001

 -- SIP/5001-87f7 is ringing

 -- SIP/5001-87f7 answered SIP/5002-a28b

 -- Attempting native bridge of SIP/5002-a28b and SIP/5001-87f7

 -- Started music on hold, class 'default', on
SIP/5002-a28b

Warning, flexibel rate not heavily tested!

Jan 30 10:26:40 NOTICE[1100258240]: res_musiconhold.c:260
monmp3thread: Request to schedule in the past?!?!

Jan 30 10:26:50 NOTICE[1234379840]: rtp.c:264
process_rfc3389: RFC3389 support incomplete. Turn off on client if possible

 -- Stopped music on hold on SIP/5002-a28b

 == Spawn extension (sip, 5001, 1) exited non-zero on
'SIP/5002-a28b'

[Asterisk-Users] call pickup

[young]

 Hi

I am using asterisk 0.7.1

 I am testing with 3 SIP phone.
 Phone A call to Phon B , and Phone B is Ringing.
 I want to pickup that call , So I press '*8' for pickup the call  on Phone
 C.

 But I can not pickup the call.

 I can see "NOTICE[6151]:chan_sip.c:5198 handle_requst: Nothing to pick up"
 in console.

;== sip.conf 
;
; SIP Configuration for Asterisk
;
[general]
port = 5060   ; Port to bind to
;bindaddr = 61.36.179.152 ; Address to bind to
bindaddr = 0.0.0.0
;externip = 200.201.202.203 ; Address that we're going to put in SIP
messages if we're behind a NAT
;localnet = 61.36.179.0  ; Internal NETWORK address
;localmask = 255.255.255.128  ; Internal netmask
;context = default  ; Default for incoming calls
context = from-sip
;srvlookup = yes  ; Enable SRV lookups on outbound calls
;pedantic = yes   ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
maxexpirey=3600  ; Max length of incoming registration we allow
defaultexpirey=160  ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes  ; Turn on support for SIP video
;disallow=all   ; Disallow all codecs
;allow=ulaw   ; Allow codecs in order of preference
;allow=ilbc
;
register => [EMAIL PROTECTED] ; Register with a SIP provider
;register => [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as
1234 here.
;
callgroup=1
pickupgroup=1


[hst239]
type=friend
secret=young
dtmfmode=inband
host=61.36.179.239
threewaycall = yes
callgroup=1
pickupgroup=1
context=from-sip

[hst220]
type=friend
host=61.36.179.220
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip

[hst238]
type=friend
host=61.36.179.238
dtmfmode=inband
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip

[hst155]
type=friend
host=210.98.251.155
callgroup=1
pickupgroup=1
threewaycall = yes
context=sip-from

[61.36.179.167]
type=friend
username=9002000
host=61.36.179.167
callgroup=1
pickupgroup=1
context=from-sip


= extensions.conf ===

[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
IAXINFO=guest ; IAXtel username/password

TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)



[iaxtel700]
exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXX,2,Congestion

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,2,Congestion

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXX,2,Congestion
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,2,Congestion
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,2,Congestion
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,2,Congestion

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
exten => 6601,1,WaitMusicOnHold(30) ; hur
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
;
; switch => IAX2/user:[EMAIL PROTECTED]/local

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used 

[Asterisk-Users] Asterisk with a laptop with built-in Intel 537 modem

[Ken Alker]

I have * working on my Sony Vaio PCG-FX120 laptop.  I am trying to get * to 
recognize my internal PCI Intel modem as an FXO port.  I have modified 
wcfxo.c in order to identify the PCI modem properly.  Based on output from 
dmesg, wcfxo didn't recognize the modem until I inserted the proper vendor 
and device IDs into wcfxo.c and re-compiled.  Note that the error message 
from modprobe wcfxo still returns No such devices, but the card IS now 
recognized based upon my dmesg results.  However, now I'm running into the 
error Out of space to write register 06 with e0 as reported by dmesg.

I tried changing ZT_CHUNKSIZE from 8 to 4 to no avail.  Per a FAQ at 
www.digium.com, my problem is that my modem shares interrupts with other 
devices on my laptop.  This is indeed the case based upon dmesg output.  It 
appears the SMbus (controller?), audio controller, and modem all share Int 
9.

Questions:
1) Is IRQ sharing really causing this problem?
2) Why can't wcfxo handle shared IRQ's, or does it not have anything to do 
with wcfxo?
3) Is there some way to force the kernel to pick a different IRQ for the 
modem?  If so, how?

Note that the BIOS for my laptop is VERY limited and appears to have no way 
to change IRQ's for devices.  I'm not very PCI literate, so perhaps (on a 
laptop), the IRQ's are hardwired and unchangable and I'm doomed entirely?

Any help or ideas would be appreciated.

---
Following are outputs of modprobe, dmesg, lspci (you may notice the 
debugging output I added to wcfxo.c where I spit out values for x, reg, and 
value as wcfxo tries to write to the modem).

[prompt]# modprobe wcfxo
/lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including 
invalid IO or IRQ parameters.
 You may find more information in syslog or the output from dmesg
/lib/modules/2.4.20-8/misc/wcfxo.o: insmod 
/lib/modules/2.4.20-8/misc/wcfxo.o failed
/lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed

[prompt]# dmesg  SNIPPET
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 9 for device 00:1f.6
PCI: Sharing IRQ 9 with 00:1f.3
PCI: Sharing IRQ 9 with 00:1f.5
x= reg=05 value=00
x=0001 reg=05 value=08
PCI: Setting latency timer of device 00:1f.6 to 64
x=0002 reg=01 value=80
x=0003 reg=07 value=00
x=0004 reg=08 value=01
x=0005 reg=09 value=89
x=0006 reg=0a value=00
x=0007 reg=05 value=0a
wcfxo: Out of space to write register 06 with e0
Failed to initailize DAA, giving up...
Zapata Telephony Interface Unloaded
[prompt]# lspci
00:00.0 Host bridge: Intel Corp. 82815 815 Chipset Host Bridge and Memory 
Controller Hub (rev 11)
00:02.0 VGA compatible controller: Intel Corp. 82815 CGC [Chipset Graphics 
Controller] (rev 11)
00:1e.0 PCI bridge: Intel Corp. 82801BAM/CAM PCI Bridge (rev 03)
00:1f.0 ISA bridge: Intel Corp. 82801BAM ISA Bridge (LPC) (rev 03)
00:1f.1 IDE interface: Intel Corp. 82801BAM IDE U100 (rev 03)
00:1f.2 USB Controller: Intel Corp. 82801BA/BAM USB (Hub #1) (rev 03)
00:1f.3 SMBus: Intel Corp. 82801BA/BAM SMBus (rev 03)
00:1f.4 USB Controller: Intel Corp. 82801BA/BAM USB (Hub #2) (rev 03)
00:1f.5 Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio 
(rev 03)
00:1f.6 Modem: Intel Corp. Intel 537 [82801BA/BAM AC'97 Modem] (rev 03)
01:00.0 FireWire (IEEE 1394): Texas Instruments TSB43AA22 IEEE-1394 
Controller (PHY/Link Integrated) (rev 02)
01:02.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev 80)
01:02.1 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev 80)
01:08.0 Ethernet controller: Intel Corp. 82801BA/BAM/CA/CAM Ethernet 
Controller (rev 03)

/**
Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU
Impulse Internet Services   http://www.impulse.net
Santa Barbara,  San Luis Obispo,  Ventura, Los Angeles, Orange
T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo
***/
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[Asterisk-Users] An out-of-band question abd Dialogic GammaLink CP4/LSI Series 2:)

[John Foster]

Dear All,

Even its not much relevant to ask at this forum but If anyone can commnet..
I m trying to run CP4/LSI on Linux RHL 8.0 box, tried with LINUX_SR5.1.tgz, LiS-2.17.A.tgz
It gives errors like..
Dialogic Shared RAM Protocol ModuleVersion 2.0Linux 2.x.xKernel 2.4.xCopyright (C) 2001 Intel Corp.ALL RIGHTS RESERVED
Unable to demand load configured STREAMS object mercdDriver, device major 233

Anyone any idea?
Someone said that this ver of dialogic card is not supported at linux at all. is there anysuch thing?

Thanks in advance
JF
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!

Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Walter Doerr]

On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote:
 hi,
 
 just signed up and it works like a charm. :-)
 They even support g711 :) and multiple channels :)
 
 make sure you have in sip.conf:
 
 register = :[EMAIL PROTECTED]/extension in your context
 

I believe that I have that entry in sip.conf. Maybe not the extension.
Still no luck.


 you will get the too many hops if you try to register
 with their proxy (proxy.de.sipgate.net).

I tried both. Besides, both names resolve to the same IP.


Here is what I just received:

Sip read: 
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 212.102.234.130:5060;branch=z9hG4bK7ba3fc35
From: asterisk sip:[EMAIL PROTECTED];tag=as2f4213d0
To: sip:217.10.79.9;tag=b11cb9bb270104b49a99a995b8c68544.566f
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 217.10.79.9:5060 Noisy feedback tells:  pid=30305
req_src_ip=217.10.79.9 req_src_port=5060 in_uri=sip:217.10.79.9
out_uri=sip:217.10.79.9 via_cnt==22


-Walter



-- 
  Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
  The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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[Asterisk-Users] Calls dropping off

[Steve Foy]

Hi,

I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.

Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.

There is nothing in the logs and nothing on the console, the call just seems
to 'go away'!

Can anyone shed any light on this?

Regards,
Steve

-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] Send DTMF tone Like 'C' on connected call

[reseaux]

Hi to all i have made a little modification to app_dial.c to play a mex when a 
call is connect like the A(mex) but from caller side option my new is B(mex). 
If someone think is good think a made patch for *. I use this mod to play a 
DTMF wav of C tone :-)
Thank in advance
Dimitri

PS:Nick how is possible?

On Friday 30 January 2004 01:15, Nick Bachmann wrote:
  Dear to all
  someone know how is possible to have a DTMF tone like C AKA Alpha
  Tone
  (connect tone) to the caller?

 Yes, it's possible.


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Re: [Asterisk-Users] Calls dropping off

[Olle E. Johansson]

Steve Foy wrote:

Hi,

I've got a fairly working Asterisk setup, with a few minor glitches, one of
which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software
SIP phones (SJPhone) connecting to * then out through analogue lines with
X100P cards.
There is nothing in the logs and nothing on the console, the call just seems
to 'go away'!
Enable 'sip debug' at the CLI and send some detailed log file.

/O

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Re: [Asterisk-Users] Echo worsens in 0.7.1

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 00:57, Eric Wieling wrote:
 Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
 being translated into Asterisk Cause Codes and being passed back to
 app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
 variable)?

HANGUPCAUSE is working fine here (cvs).

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)

iD8DBQFAGkv82TEAILET3McRAjGcAJ9FzGmcXX8jJwjs30hVjhAO3pcO5ACfZ6mr
pRRyhh0J/GeyezwX1m8Qi1s=
=PbAl
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Walter Doerr]

On Fri, Jan 30, 2004 at 12:22:18PM +0100, Walter Doerr wrote:
 On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote:
  hi,
  
  just signed up and it works like a charm. :-)
  They even support g711 :) and multiple channels :)
  
  make sure you have in sip.conf:
  
  register = :[EMAIL PROTECTED]/extension in your context
  
 
 I believe that I have that entry in sip.conf. Maybe not the extension.
 Still no luck.
 

Following up to my own message:

* is working with sipgate now (should be no surprise as they are using
* too).

Apparently I have no idea how to setup a sip.conf file.

I have the above mentioned register command and in addition a
[sipgate] section in sip.conf.
After removing the [sipgate] section everything works fine.


-Walter



-- 
  Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
  The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] HANGUPCAUSE

[CW_ASN - Gus]

How? Is written in CDR?

Regards,

Gus

- Original Message - 
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 00:57, Eric Wieling wrote:
 Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
 being translated into Asterisk Cause Codes and being passed back to
 app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
 variable)?

HANGUPCAUSE is working fine here (cvs).

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)

iD8DBQFAGkv82TEAILET3McRAjGcAJ9FzGmcXX8jJwjs30hVjhAO3pcO5ACfZ6mr
pRRyhh0J/GeyezwX1m8Qi1s=
=PbAl
-END PGP SIGNATURE-

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Re: [Asterisk-Users] HANGUPCAUSE

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
 HANGUPCAUSE is working fine here (cvs).
 How? Is written in CDR?

CDRs contain BUSY when busy and NO ANSWER on the rest.

extensions.conf:

[provider-out]
...
exten = _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten = _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)

[provider-out-failed]
exten = c1,1,Hangup()

exten = c2,1,Busy()

exten = c3,1,Answer()
exten = c3,2,ResetCDR()
exten = c3,3,Playtones(info)
exten = c3,4,Wait(60)
exten = c3,5,Hangup()

exten = c4,1,Congestion()

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)

iD8DBQFAGlKy2TEAILET3McRAv7gAKCREpAN3kVvbEuTDAQkU9kb6IrZiQCdEXlR
3FroTgPgWQmBrqGwjwktmvc=
=yyxo
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[Asterisk-Users] billing software

[Deepakumar JV]

Hello

Is anyone using a commercial billing 
software with *  which product is that?

i am looking for using with pre-paid as 
well as post paid.

Also where can i find info about voip 
regulation/licenses to become a provider???


Thanks
Deepak

[Asterisk-Users] ZAPRTC load error

[info-lists]

I have compiled the zaptel library and zaprtc on a system that gives the
following from uname -a:
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC
2002 i686 unknown

Makefile for zaptel had the following line uncommented:
#
KFLAGS+=-D__SMP__


When doing the make load for zaprtc I get the following error:

modprobe zaptel
/lib/modules/2.4.18-64GB-SMP/misc/zaptel.o: kernel-module version mismatch
/lib/modules/2.4.18-64GB-SMP/misc/zaptel.o was compiled for kernel
version 2.4.18-4GB
while this kernel is version 2.4.18-64GB-SMP.

Any ideas on where to look for the solution would be appreciated.  Have
checked the Makefiles but didn't see anything related.

Thanks,
Robert



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[Asterisk-Users] mediatrix, dtmf

[Dawid Mielnik]

Hi,

I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104
FXS. I can not enter mailbox number (voicemail) or pin code (meet-me).
Asterisk shows 'username not entered' when dialing in voicemail.

Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ?

Best regards,

Dave

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Re: [Asterisk-Users] Calls dropping off

[Bill Hamel]

Shot in the dark here ...

Do you have: 

canreinvite=no

Set in sip.conf for the SIP phones in question ?

Ciao,
-b


Quoting Steve Foy [EMAIL PROTECTED]:

 Hi,
 
 I've got a fairly working Asterisk setup, with a few minor glitches, one of
 which is very very irritating.
 
 Sometimes, during a call, the remote end just drops off. We're using
 software
 SIP phones (SJPhone) connecting to * then out through analogue lines with
 X100P cards.
 
 There is nothing in the logs and nothing on the console, the call just seems
 to 'go away'!
 
 Can anyone shed any light on this?




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[Asterisk-Users] Auto dial in Off Hook situation.

[David J Carter]

Hi all,


I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.


Anyone done this or know how it's done.


regards


Dave

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Re: [Asterisk-Users] * with OH323 - Memory Leak

[Michael Manousos]

Todd Wallace wrote:
I noticed in the BUGS that there is a memory leak with * using
asterisk-oh323.  If we use SIP primarily as the main protocol, but OH323 on
occasion to test some international routes on our Nextone MSW...How bad is
the Memory leak that is described??
Todd Wallace


This was a bug of the early releases of asterisk-oh323. It's been
fixed in current release (forgot to mark it as FIXED).
Michael.

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[Asterisk-Users] P2P RTP without SIP re-invites

[Low, Adam]

I'm confronted with an issue that I am sure many others are too with Asterisk and 
scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a 
large volume of simultaneous calls but have the feeling that the hardware requirements 
to handle large volumes of RTP streams would be too vast.

So with that assumption I imagine a platform that would not get involved with the 
actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each 
end of the call deal with RTP encoding with their dedicated DSP hardware. There is an 
alternative in mind that maybe I could utilise some old Dialogic DSP cards that we 
have but I suspect trying to get these working with Asterisk would be a lot of 
programming work that I probably couldn't manage, maybe I'm wrong ?

The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd 
(specifically SIP breaks and calls are not torn down correctly) and of course you lose 
a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when 
it is in the SIP signalling path.

I vaguely remember previous discussions on this and even a patch but I am unable to 
find anything in the archives, does anybody have any info on that ?

The conclusion I have come to is that I would try and patch the Asterisk code. The 
idea being that when the RTP parameters are negotiated that Asterisk would pass 
through the source address/port from each SIP client causing them to talk RTP 
directly. I intend to begin work on this this weekend but am I hoping that maybe 
somebody else has already achieved what I desire, anybody ?

Rgds,
Adam




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[Asterisk-Users] IAX call problems

[Rattana BIV]

hi,

I use IAX softphone with asterisk and I notice that 
a call between two IAX softphonesend after 1 min. Then I can't hear 
anything but the call still in progress.
I have this log in asterisk IAX debug:

Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: 
IAX Subclass: 
ACK Timestamp: 
00016ms SCall: 21589 DCall: 1 
[192.168.1.22:4569] 
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE 
Subclass: 2 
Timestamp: 65795ms SCall: 6 DCall: 21588 
[192.168.1.22:4569] 
Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE 
Subclass: 2 
Timestamp: 65795ms SCall: 6 DCall: 21588 
[192.168.1.22:4569] 
Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: 
IAX Subclass: 
ACK Timestamp: 
65795ms SCall: 21588 DCall: 6 
[192.168.1.22:4569] 
Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: 
IAX Subclass: 
PING Timestamp: 
75906ms SCall: 22105 DCall: 5 
[192.168.1.77:4569] 
Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: 
IAX Subclass: 
ACK Timestamp: 75906ms 
SCall: 5 DCall: 22105 [192.168.1.77:4569]


Any suggestions ???


Thanks in advance

Rattana


PS: The softphone I use work with wiax.dll and is 
developpe by me =)

Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Philipp von Klitzing]

Hi!

 just signed up and it works like a charm. :-)
 They even support g711 :) and multiple channels :)

 make sure you have in sip.conf:
 register = :[EMAIL PROTECTED]/extension in your context

Their tech support just told me that it takes a while until a registered
user becomes available to the signaling server. I tried yesterday, SIP
registration worked fine, but dialing failed with 404 (not found) or 484
(address incomplete).

Anyway, the nice thing is that you can use ENUM to call sipgate users.
And they have a freely accessible FWD and IAXtel gateway as well in D
(Düsseldorf) and UK.

Cheers, Philipp


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Re: [Asterisk-Users] Calls dropping off

[Steve Foy]

On Fri, Jan 30, 2004 at 01:18:29PM +0100, Olle E. Johansson wrote:
 Enable 'sip debug' at the CLI and send some detailed log file.

It's very difficult to catch the logs when this happens, it doesn't happen
all the time, and I'm hardly ever on the phone so, it would be even less
likely to happen to me.

Is there a way I can get the sip debug lines to get piped out to a file with
timestamps?

-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] Expire old voice mail messages, et al

[David Gomillion]

Jeff Crews wrote:
[snip]
 Any thought of having maximum number of messages be defined globally
 in voicemail.conf or on a per user basis?


I think this is a good idea.  But instead of the two extremes, maybe we
could come up with a class of service definition (idea shamelessly stolen
from Nortel).  That way, we could define how long each of the message
parameters can be, i.e. how many MB of messages, how big of greetings, etc.

 Also, does anyone feel a need to have the voicemail system speak the
 date and time the voice mail message arrived for those that access
 messages by phone instead of the usual email?


Yes.  We need that.  And it seems to work on last testing, if we set the
time zone.  I'll have to check it again.

 Finally...am I the only person who does not have a need for separate
 busy and no answer outgoing messages?  When I change my greeting...I
 change the not available...and have a cron job copy the unavailable
 to the busy file so the messages are the same.

If you have no need for different messages, then change your
extensions.conf, and set them both to go to (u), instead of one going to
(b) and the other going to (u).

Hope this helps,
David Gomillion

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Re: [Asterisk-Users] Calls dropping off

[Steve Foy]

Bill,

On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
 Shot in the dark here ...
 
 Do you have: 
 
 canreinvite=no
 
 Set in sip.conf for the SIP phones in question ?

No, I don't.

All I have in sip.conf is the general stuff like:

   [general]
   port = 5060   ; Port to bind to (SIP is 5060)
   bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)

   allow=all
   allow=GSM
   allow=G729
   allow=iLBC
   allow=SpeeX; Allow all codecs
   allow=ulaw

and then about 10 friends like this:

   ; Shirley
   [100]
   type=friend
   username=xxx
   secret=xxx
   host=dynamic
   dtmfmode=rfc2833
   callerid=Shirley O'Neill 100
   context=internal
   [EMAIL PROTECTED]
   qualify=yes

-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] asterisk with big number of extentions.

[David Gomillion]

Chris Albertson wrote:
 What are you talking about?  You can already reload the dialplan
 without affecting existing calls.

[snip]
 But for my use this is not important as at most I'd be running
 a small office with ~10 lines were people go home at night.
 But with 25,000 active users ...when convenient would be a long,
 long time.


What we've got here is... failure to communicate. - GnR, Civil War

The when convenient is usually tied to restart.  Reload reloads
configuration files.  Restart now will, indeed, drop conversations, while
reload will not.  Reload will not change everything, so for some changes,
you will have to issue a restart; however, I have been successful in
changing my dialplan in extensions.conf (and it's related #include'd files)
and reloaded to see the changes immediately.



 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

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[Asterisk-Users] Firefly and asterisk*

[FastJack]

GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy:
just add an extention

exten = _8XXX,1,Answer
exten = _8XXX,2,DigitTimeout,5
exten = _8XXX,3,ResponseTimeout,10
exten =
_8XXX,4,Dial,IAX2/*YOUR_FIREFLY_NUMBER*:[EMAIL PROTECTED]
.com/${EXTEN}|60|T

now I can call users in the firefly-network from every phone that is
connected to my asterisk*-box.

I also added a register = ... entry to my iax.conf but this doesn't seam to
work.

Anyone knows how to receive calls on my asterisk*-box from the
firefly-network?

thanks!

- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 10:12 AM
Subject: Re: [Asterisk-Users] Introducing Firefly


Hi,

I downloaded this the other day and finally got it to stop crashing. It
appears that any response from asterisk
that implies an error (for example dialing a non-existant number, using the
wrong password, selecting a codec
that you've configured a local * not to use etc) resulted in a crash. I've
only tested the IAX proto not sip or your
own network. running XP with uptodate patches on a local lan.

When it works it works really well, although I don;t particularly like in
initial beep and end beep when i make
a call (I haven't played with all the options so it may be that I can turn
this off).. sound quality is good. All in all
a nice little app. Are you planning on allowing other people to run your
server side (like Jabber does) in their
environments?

If you need any further debugging info on the crashes, let me know...

HTH

Andy


*** REPLY SEPARATOR  ***

On 28/01/2004 at 12:11 Adam Hart wrote:

After many months of development, I'm pleased to announced Firefly - an
IAX soft phone and network.

The firefly softphone - free, runs under windows, allows connection to
multiple networks, skinable interface, connection to firefly network, IAX2
protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw,
GSM. - contact lists, selectable ringtones.

download from here - http://www.virbiage.com/firefly/

The firefly network - also free, runs on an enhanced version of IAX2
(simply uses IAX2 text messages for customised part), voicemail, text
messaging, online presence, ability to indicate status (available, away,
NA). I believe you can connect using a standard asterisk box but you'll
miss out on the extended features. The network runs on iLBC so
unforunately it won't work with most IAX2 clients (unless you get * to
translate)

Thousands of people have used it but it's still regarded in beta, as we
are still in heavy development (so expect a few bugs). It doesn't use
iaxcomm as we needed our own framework to support sip, including our own
jitterbuffer. If you don't wish to connect to the firefly network, click
cancel when it asks you.

Coming soon features
SIP - in alpha, few bugs outstanding
music onhold - playing mp3s while the other party is onhold
fast audio - will reduce the latency by 40-50ms
speex - (if anyone wants it?)

Feel free to contact me on or off the list to report bugs and suggestions.
I'll post everytime we release a new version (probably every week),
including fixed bugs and new features

Our website is http://www.virbiage.com/


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Re: [Asterisk-Users] X100P limit per PC

[David Gomillion]

Isamar Maia wrote:
 I know that it was commented here already but how many X100Ps
 I can plug per PC?

How many PCI slots do you have?  How many IRQ's can your BIOS allow you to
assign?  There is not a hard and fast rule, as far as I can tell, but these
questions may give you an idea of how many you can hope for...


 Isamar


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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[John Todd]

[This starts to look very off-topic for the -users list, which is why
I've proposed on several occasions a -biz list, but since there is no
-biz list, I'll continue this thread here.]
I currently do not have German as a language at my command, and there
are no English translations on the sipgate pages.  I am wondering if
one of the more bi-lingual members of the list here could translate
the product offerings for me.  I am looking for a phone number in .de
that will map by ENUM (and of course, by PSTN-to-SIP) to one of my *
servers.
I have family who communicate regularly with people in .de and other
nations in Western Europe, and I'd like to give them the people in
Europe who are not Internet-savvy a way to reach us without overseas
toll charges.  At the same time, I'd also like to have an EU
presence for my own reasons.
If someone could provide the quick summary and costs for such a
service from sipgate, that would be great.
JT


hi,

just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)
make sure you have in sip.conf:

register = :[EMAIL PROTECTED]/extension in your context

you will get the too many hops if you try to register
with their proxy (proxy.de.sipgate.net).
best regards

kapejod
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


 I set up an account with sipgate yesterday evening and tried to use the
 above mentioned register in sip.conf * to login to sipgate.
 No luck so far.
 They use SER and I get 483 too many hops replies back from them.

 Any help is greatly appreciated.

 -Walter



 --
   Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
   The poor folks who only have 100MBytes of RAM five years
 from now may not be able to buffer a 16MB packet, but that's their tough
 luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] Auto dial in Off Hook situation.

[John Todd]

Hi all,

I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.
Anyone done this or know how it's done.

regards

Dave
Depends on the phone.  If you have an FXS interface, look for 
immediate= in your zapata.conf file.

If you have an IP phone, search the vendor's documentation for PLAR 
(Private Line Auto Ringdown)

JT
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[Asterisk-Users] Re: Grandstream Firmware ?

[Stephen R. Besch]

Greg Boehnlein wrote:

On Thu, 29 Jan 2004, Michael Welter wrote:


I have 1.0.4.45 (beta) on my tftp server.  Try it at 66.250.23.58.

Cheers,
Michael Welter


Is there a changelog available for the Beta release train? I'm looking to 
see if they have fixed Early Dial yet.

When GS connected to my * server to examine the problem, they promised 
to keep me posted on the early dial problem.  I haven't heard anything 
yet, so I am assuming that it has not been fixed.

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Re: [Asterisk-Users] asterisk with big number of extentions.

[John Todd]

At 8:16 AM -0600 1/30/04, David Gomillion wrote:
[snip]
The when convenient is usually tied to restart.  Reload reloads
configuration files.  Restart now will, indeed, drop conversations, while
reload will not.  Reload will not change everything, so for some changes,
you will have to issue a restart; however, I have been successful in
changing my dialplan in extensions.conf (and it's related #include'd files)
and reloaded to see the changes immediately.
Try extensions reload to load up just those changes you've made in 
the extensions.conf and related #include files.

JT
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Re: [Asterisk-Users] HANGUPCAUSE

[CW_ASN - Gus]

Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .

Regards,

Gus

- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:48 AM
Subject: Re: [Asterisk-Users] HANGUPCAUSE


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
 HANGUPCAUSE is working fine here (cvs).
 How? Is written in CDR?

CDRs contain BUSY when busy and NO ANSWER on the rest.

extensions.conf:

[provider-out]
...
exten = _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten = _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)

[provider-out-failed]
exten = c1,1,Hangup()

exten = c2,1,Busy()

exten = c3,1,Answer()
exten = c3,2,ResetCDR()
exten = c3,3,Playtones(info)
exten = c3,4,Wait(60)
exten = c3,5,Hangup()

exten = c4,1,Congestion()

- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)

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3FroTgPgWQmBrqGwjwktmvc=
=yyxo
-END PGP SIGNATURE-

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[Asterisk-Users] IAX1 vs IAX2 for IAXtel

[Vic Cross]

G'day list,

I am getting a lot[1] of traffic on my Internet link, ICMP messages from
69.73.19.178 telling me UDP port 5036 is unreachable (this IP address 
belongs to iaxtel.org).

I see from the wiki that IAXtel supports only IAX2 from December 2003.  
Fine, however it looks like my * still wants to try and register using 
IAX1, and I can't find how to turn this off.

This situation is confirmed in the response from iax show registry and 
iax2 show registry:

enterprise*CLI iax show registry
HostUsername Perceived  Refresh  State
69.73.19.178:5036   ##myid## Unregistered  60  Request Sent
enterprise*CLI iax2 show registry
HostUsername Perceived  Refresh  State
69.73.19.178:4569   ##myid## ##myIP##:4569   60  Registered

Can I shut off these attempts to register using IAX1 (please forgive me if 
it's obvious, but I've been through the wiki and the Handbook)?

Cheers,
Vic Cross

[1] Okay, so it's not actually a lot of data, but it's four 
request/responses every ten seconds, non-stop...
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[Asterisk-Users] G729 license

[Jess Magnaye]

Hello all,

I would like to just verify where to purchase the 
G729 license for Asterisk. Like I want to run G729 codec for all my calls 
passing thru Asterisk (voicemail, parking, via ZAP, via SIP, etc). The 
list says license is taken from Digium, does that apply also if I have Dialogic 
cards on my *?

[Asterisk-Users] newb info needed

[Jeff Donovan]

greetings

I am interested in building asterisk on a BSD/ OSX platform.
is there a source i can compile?
for testing purposes would i be able to use a modem for outside line 
connections?
any info would be helpful

thanks

--jeff

---
jeff donovan
basd network operations
(610) 807 5571 x4
AIM  xtdonovan 

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[Asterisk-Users] PHP developer Wanted ! :-)

[reseaux]

Dear ALL
i need to develop a web frontend for my * app i need only manage data from 
MySQL db, i will pay to develop it (not much :-) )
Thanks in advance
Dimitri

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Re: [Asterisk-Users] cdr_addon_mysql compile error

[Asterisk VOIP]

On Thu, 29 Jan 2004 20:01:05 -0600, Tilghman Lesher wrote
 On Thursday 29 January 2004 19:40, Asterisk VOIP wrote:
  almost got it.  I now get the following in the CLI,
   ERROR[1226054960]: cdr_addon_mysql.c:203 mysql_log: Failed to insert
  into database.  db is setup correctly.
 
 You probably have a single field missing or misspelled.  Recheck your
 table definition.
 
  asterisk is started with /usr/sbin/asterisk -c but the only
  messages I find are, LOG_ERROR, in the CLI.  Where are the LOG_DEBUG
  messages?
 
 They aren't actually logged to disk or to the CLI unless you 
 configure them to go there.  Look in /etc/asterisk/logger.conf.
 
 -Tilghman
 
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got it!  last question the uniqueid and userfield info
is empty for all calls.  should I Add a CFLAGS+=-DMYSQL_LOGUNIQUEID to the
Makefile for cdr_addon_mysql.c and do a make clean make install to get the
uniqueid info?  what about the userfield info?

thanks for all of your help.

-wr


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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Philipp von Klitzing]

Hi!

 the product offerings for me.  I am looking for a phone number in .de
 that will map by ENUM (and of course, by PSTN-to-SIP) to one of my *
 servers.

Both nikotel.de and sipgate.de offer such a number. With Nikotel you'll
need to spend at least 7 € per month for such a number (which is a very
good deal). I don't think they run ENUM.

With sipgate.de this in-dialing number is for free - currently you can
have a number in Düsseldorf (D), capital of the state Nordrhein-
Westfalen aka the biggest state in the union, 0211 area code, or London
(UK) or Reading (UK), more German cities will follow soon. They reserve
the right to cut the free services at any time or start charging for them
- who wouldn't have guessed that; they remain owner of the in-dialing
phone number. Their FAQ says they are planning on IAX support in the near
future. IP-2-phone tariffs will follow in February. They are running SER.

  search = enum1.sipgate.net
  search = enum2.sipgate.net

If you check the persons in charge you'll find that this company is
closely linked with www.netzquadrat.de.

 I have family who communicate regularly with people in .de and other
 nations in Western Europe, and I'd like to give them the people in
 Europe who are not Internet-savvy a way to reach us without overseas
 toll charges.

You could also use their FWD or IAXtel gateway:

enter at +49 211 58000 100
or at +44 20 7127 6200

and then dial

   IAXTel (000700)
   Freeworlddialup (000393)
   iptel (000477)
   Sipphone (000747)

  At the same time, I'd also like to have an EU presence for my own
 reasons.

Hehe - for the .eu domain you'll have to wait until October or so. But I
guess that's not what you meant. ;-

Cheers, Philipp


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Re: [Asterisk-Users] HANGUPCAUSE

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
 Ok, but is not working as expected... we can't see clear ISUP causes. We
 can't make different treatments or store other causes than busy (cause=17)
 in cdr's .

You could use my approach and combine it with the CDR userfield. Personally I 
would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)

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5B+arXbMx37BtKSFLez3KlI=
=61o0
-END PGP SIGNATURE-

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Re: [Asterisk-Users] IAX1 vs IAX2 for IAXtel

[Mark Spencer]

in the *short term* just add:

noload = chan_iax.so

to your modules.conf

Eventually we will move chan_iax2.c to chan_iax.c and chan_iax.c will
become chan_iax1.c and will likely not be a default part of the build
process.

Mark

On Sat, 31 Jan 2004, Vic Cross wrote:

 G'day list,

 I am getting a lot[1] of traffic on my Internet link, ICMP messages from
 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address
 belongs to iaxtel.org).

 I see from the wiki that IAXtel supports only IAX2 from December 2003.
 Fine, however it looks like my * still wants to try and register using
 IAX1, and I can't find how to turn this off.

 This situation is confirmed in the response from iax show registry and
 iax2 show registry:

 enterprise*CLI iax show registry
 HostUsername Perceived  Refresh  State
 69.73.19.178:5036   ##myid## Unregistered  60  Request Sent
 enterprise*CLI iax2 show registry
 HostUsername Perceived  Refresh  State
 69.73.19.178:4569   ##myid## ##myIP##:4569   60  Registered

 Can I shut off these attempts to register using IAX1 (please forgive me if
 it's obvious, but I've been through the wiki and the Handbook)?

 Cheers,
 Vic Cross

 [1] Okay, so it's not actually a lot of data, but it's four
 request/responses every ten seconds, non-stop...
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Re: [Asterisk-Users] How to delay dialing

[Walker Haddock]

On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
 Hi there,
 
 I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO
 line.  The reason being is that Voicetronix sends out the DTMF too fast even
 before the line is fully established with the carrier.  Usually when dialing
 an 8 digit number, only 7 digits are actually successfully heard by the
 carrier.
 
 Currently, my dial plan is:
 exten = _9.,1,Dial(vpb/1-1/${EXTEN:1})
 
 Daniel said to insert a , before the numbers.  I am not too sure where to
 insert it.  I tried
 exten = _9.,1,Dial(vpb/1-1/,${EXTEN:1}) and that seems to be cause a
 parsing error.
 
 Anybody has any ideas for a hack?
 
 David
The token to insert a pause is `W` (must be upper case).  Try this:

 exten = _9.,1,Dial(vpb/1-1/W${EXTEN:1})

each `W` will cause a 0.5 second pause.

-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
***
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Thilo Salmon]

John,

at the moment  German and UK geographic numbers are available free of
charge with unlimited inbound traffic through SIP. ENUM mappings are
static in the way that enum zones cannot be configured by a user, but
point to a SIP UAS at which you can register an * box. Let me know, if
you want me to sign up. 

How is freenum.org taking on?

Thilo

On Fri, 2004-01-30 at 15:21, John Todd wrote:
 [This starts to look very off-topic for the -users list, which is why 
 I've proposed on several occasions a -biz list, but since there is no 
 -biz list, I'll continue this thread here.]
 
 I currently do not have German as a language at my command, and there 
 are no English translations on the sipgate pages.  I am wondering if 
 one of the more bi-lingual members of the list here could translate 
 the product offerings for me.  I am looking for a phone number in .de 
 that will map by ENUM (and of course, by PSTN-to-SIP) to one of my * 
 servers.
 
 I have family who communicate regularly with people in .de and other 
 nations in Western Europe, and I'd like to give them the people in 
 Europe who are not Internet-savvy a way to reach us without overseas 
 toll charges.  At the same time, I'd also like to have an EU 
 presence for my own reasons.
 
 If someone could provide the quick summary and costs for such a 
 service from sipgate, that would be great.
 
 JT
 
 
 hi,
 
 just signed up and it works like a charm. :-)
 They even support g711 :) and multiple channels :)
 
 make sure you have in sip.conf:
 
 register = :[EMAIL PROTECTED]/extension in your context
 
 you will get the too many hops if you try to register
 with their proxy (proxy.de.sipgate.net).
 
 
 best regards
 
 kapejod
 --
 Klaus-Peter Junghanns
 
 CEO, CTO
 Junghanns.NET GmbH
 Breite Straße 13 - 12167 Berlin - Germany
 fon: (de) +49 30 79705390
 fon: (uk) +44 870 1244692
 fax: (de) +49 30 79705391
 iaxtel: 1-700-157-8753
 http://www.Junghanns.NET/asterisk/
 
 
 
   I set up an account with sipgate yesterday evening and tried to use the
   above mentioned register in sip.conf * to login to sipgate.
   No luck so far.
 
   They use SER and I get 483 too many hops replies back from them.
 
   Any help is greatly appreciated.
 
   -Walter
 
 
 
   --
 Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
 The poor folks who only have 100MBytes of RAM five years
   from now may not be able to buffer a 16MB packet, but that's their tough
   luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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40233 Duesseldorf|  Fax: +49 211 302033 22


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Re: [Asterisk-Users] cdr_addon_mysql compile error

[Tilghman Lesher]

On Friday 30 January 2004 09:41, Asterisk VOIP wrote:
 On Thu, 29 Jan 2004 20:01:05 -0600, Tilghman Lesher wrote

  On Thursday 29 January 2004 19:40, Asterisk VOIP wrote:
   almost got it.  I now get the following in the CLI,
ERROR[1226054960]: cdr_addon_mysql.c:203 mysql_log: Failed to
   insert into database.  db is setup correctly.
 
  You probably have a single field missing or misspelled.  Recheck
  your table definition.
 
   asterisk is started with /usr/sbin/asterisk -c but the only
   messages I find are, LOG_ERROR, in the CLI.  Where are the
   LOG_DEBUG messages?
 
  They aren't actually logged to disk or to the CLI unless you
  configure them to go there.  Look in /etc/asterisk/logger.conf.

 got it!  last question the uniqueid and userfield info
 is empty for all calls.  should I Add a
 CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile for cdr_addon_mysql.c
 and do a make clean make install to get the uniqueid info?

If you want that, yes.

 what about the userfield info?

It only gets set if you execute SetCDRUserField() during the dialplan.
Obviously, what you choose to put in the userfield is user-defined
(i.e. the default is always going to be blank).

-Tilghman

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[Asterisk-Users] Address Separator hex b causes callerid rejection

[bam]

I am having a little bit of a problem with BT rejecting my callerid values 
as they are prefixed by hex b. This indicates that the caller id is user 
provided and not verified.

Does anyone know how I can control where this appears in the cli?

The purpose of the separator is described below:

1 - PNO 006 section 2.4.19 c note states that the hex b denotes an
address separator, to separate the part which is network provided from
that which is user provided - This means that it separates the extension
number from the rest of the number.
2 - PNO008 section 22.1.3.3 states that the hex b dependant on its
position, denotes whether the screening indicator is user provided not
verified, network provided or user provided verified and passed.
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[Asterisk-Users] Adtran 750 DID question.

[Bisker, Scott (7805)]

Hello All,

I've mostly solved my DID problem from a few days ago.  Apparenly the lines weren't 
configured properly.  Now heres the next question.  12 EM wink lines from telco.  I 
have them all plugging into an Adtran 750 with FXS cards.  The Adtran ports are 
configured DPO.   How do I signal this from Zaptel.  I have them setup EM in 
zaptel.conf and EM_W in zapata.conf.  They work, however, no DNIS info is being 
passed.  Do I need to signal these something different like loopstart or kewlstart, so 
the DNIS info gets passed?  I watch the Tx/Rx bits from zttool, and everything looks 
okay coming from the Adtran.  It looks like asterisk isn't winking properly.

When I had the lines misconfigured for fxs_ls the DNIS info was passing fine.

I'm running 
zaptel-0.8.0
libpri-0.5.1
And asterisk CVS from 12/23/2003
RedHat 8.0
Dual 2.4 Xeon Processors (hyperthreading disabled)
2Gig Memory

Any help would be greatly appreciated.

Regards,

-sb
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Re: [Asterisk-Users] Music on Hold Warnings

[Tilghman Lesher]

On Friday 30 January 2004 04:33, Craig Waddington wrote:
 1.Warning, flexibel rate not heavily tested!

You're using variable rate mp3's.  If you want to avoid the error,
recode your mp3s to a static rate.

 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request
 to schedule in the past?!?!

Is your machine heavily loaded?  This could indicate that a thread was
unable to complete a task because it was interrupted and did not
resume for a fairly long time (as processor time goes).  It could also
indicate clock drift (sync your time with NTP servers more often).

-Tilghman

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Re: [Asterisk-Users] ZAPRTC load error

[Tilghman Lesher]

On Friday 30 January 2004 07:04, [EMAIL PROTECTED] wrote:
 I have compiled the zaptel library and zaprtc on a system that
 gives the following from uname -a:
 Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27
 13:58:12 UTC 2002 i686 unknown

 Makefile for zaptel had the following line uncommented:
 #
 KFLAGS+=-D__SMP__


 When doing the make load for zaprtc I get the following error:

 modprobe zaptel
 /lib/modules/2.4.18-64GB-SMP/misc/zaptel.o: kernel-module version
 mismatch /lib/modules/2.4.18-64GB-SMP/misc/zaptel.o was compiled
 for kernel version 2.4.18-4GB
 while this kernel is version 2.4.18-64GB-SMP.

 Any ideas on where to look for the solution would be appreciated. 
 Have checked the Makefiles but didn't see anything related.

Recompile your kernel or install the kernel source which exactly
matches the running kernel, then recompile and reinstall zaptel.

-Tilghman

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Re: [Asterisk-Users] IAX1 vs IAX2 for IAXtel

[Tilghman Lesher]

On Friday 30 January 2004 08:59, Vic Cross wrote:
 G'day list,

 I am getting a lot[1] of traffic on my Internet link, ICMP messages
 from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP
 address belongs to iaxtel.org).

 I see from the wiki that IAXtel supports only IAX2 from December
 2003. Fine, however it looks like my * still wants to try and
 register using IAX1, and I can't find how to turn this off.

bash# touch /etc/asterisk/iax1.conf
bash# asterisk -rx reload

This will pretty much turn off chan_iax without unloading it.

-Tilghman

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Re: [Asterisk-Users] Calls dropping off

[Bill Hamel]

Try adding it to the phones involved so it looks like this:

; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
canreinvite=no

-b


Quoting Steve Foy [EMAIL PROTECTED]:

 Bill,
 
 On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
  Shot in the dark here ...
  
  Do you have: 
  
  canreinvite=no
  
  Set in sip.conf for the SIP phones in question ?
 
 No, I don't.
 
 All I have in sip.conf is the general stuff like:
 
[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
 
allow=all
allow=GSM
allow=G729
allow=iLBC
allow=SpeeX; Allow all codecs
allow=ulaw
 
 and then about 10 friends like this:
 
; Shirley
[100]
type=friend
username=xxx
secret=xxx
host=dynamic
dtmfmode=rfc2833
callerid=Shirley O'Neill 100
context=internal
[EMAIL PROTECTED]
qualify=yes
 
 -- 
 Steve Foy|  http://www.unite.net
 UNITE Solutions  |  Tel: 028 9077 7338 
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Re: [Asterisk-Users] Adtran 750 DID question.

[James Sharp]

 Hello All,

 I've mostly solved my DID problem from a few days ago.  Apparenly the
 lines weren't configured properly.  Now heres the next question.  12 EM
 wink lines from telco.  I have them all plugging into an Adtran 750 with
 FXS cards.  The Adtran ports are configured DPO.   How do I signal this
 from Zaptel.  I have them setup EM in zaptel.conf and EM_W in
 zapata.conf.  They work, however, no DNIS info is being passed.  Do I need
 to signal these something different like loopstart or kewlstart, so the
 DNIS info gets passed?  I watch the Tx/Rx bits from zttool, and everything
 looks okay coming from the Adtran.  It looks like asterisk isn't winking
 properly.

I had a similar problem.  I ended up setting the trunks to either just
plain em or featd (I don't remember).  I chased through the chan_zap
source code and decided (maybe incorrectly) that asterisk doesn't look for
DNIS digits in EM Wink mode.
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Thilo Salmon]

Oops, wrong email, please ignore.

Thanks,

Thilo

On Fri, 2004-01-30 at 17:16, Thilo Salmon wrote:
 John,
 
 at the moment  German and UK geographic numbers are available free of
 charge with unlimited inbound traffic through SIP. ENUM mappings are
 static in the way that enum zones cannot be configured by a user, but
 point to a SIP UAS at which you can register an * box. Let me know, if
 you want me to sign up. 
 
 How is freenum.org taking on?
 
 Thilo
 
 On Fri, 2004-01-30 at 15:21, John Todd wrote:
  [This starts to look very off-topic for the -users list, which is why 
  I've proposed on several occasions a -biz list, but since there is no 
  -biz list, I'll continue this thread here.]
  
  I currently do not have German as a language at my command, and there 
  are no English translations on the sipgate pages.  I am wondering if 
  one of the more bi-lingual members of the list here could translate 
  the product offerings for me.  I am looking for a phone number in .de 
  that will map by ENUM (and of course, by PSTN-to-SIP) to one of my * 
  servers.
  
  I have family who communicate regularly with people in .de and other 
  nations in Western Europe, and I'd like to give them the people in 
  Europe who are not Internet-savvy a way to reach us without overseas 
  toll charges.  At the same time, I'd also like to have an EU 
  presence for my own reasons.
  
  If someone could provide the quick summary and costs for such a 
  service from sipgate, that would be great.
  
  JT
  
  
  hi,
  
  just signed up and it works like a charm. :-)
  They even support g711 :) and multiple channels :)
  
  make sure you have in sip.conf:
  
  register = :[EMAIL PROTECTED]/extension in your context
  
  you will get the too many hops if you try to register
  with their proxy (proxy.de.sipgate.net).
  
  
  best regards
  
  kapejod
  --
  Klaus-Peter Junghanns
  
  CEO, CTO
  Junghanns.NET GmbH
  Breite Straße 13 - 12167 Berlin - Germany
  fon: (de) +49 30 79705390
  fon: (uk) +44 870 1244692
  fax: (de) +49 30 79705391
  iaxtel: 1-700-157-8753
  http://www.Junghanns.NET/asterisk/
  
  
  
I set up an account with sipgate yesterday evening and tried to use the
above mentioned register in sip.conf * to login to sipgate.
No luck so far.
  
They use SER and I get 483 too many hops replies back from them.
  
Any help is greatly appreciated.
  
-Walter
  
  
  
--
  Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
  The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their tough
luck.  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Ronsdorfer Str. 74   |  Fon: +49 211 302033 12
40233 Duesseldorf|  Fax: +49 211 302033 22


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Re: [Asterisk-Users] HANGUPCAUSE

[Eric Wieling]

Personally I would like AST_CAUSE to be the Asterisk cause code (which
should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE,
SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people.

On Fri, 2004-01-30 at 09:57, Tais M. Hansen wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
  Ok, but is not working as expected... we can't see clear ISUP causes. We
  can't make different treatments or store other causes than busy (cause=17)
  in cdr's .
 
 You could use my approach and combine it with the CDR userfield. Personally I 
 would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.
 
 - -- 
 Regards,
 Tais M. Hansen
 ComX Networks
 Tel: +45-70257474
 Fax: +45-70257374
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.2.3 (GNU/Linux)
 
 iD8DBQFAGn8E2TEAILET3McRAuk4AJ4ljoWNtJSg/aPUOuodWwiC/MA1aQCgg/EG
 5B+arXbMx37BtKSFLez3KlI=
 =61o0
 -END PGP SIGNATURE-
 
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Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
Asterisk Resource Pages.

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] How to delay dialing

[Eric Wieling]

Are you sure this works for VoiceTronix Driver?  It's not implemented in
app_dial, but in chan_zap.

On Fri, 2004-01-30 at 10:15, Walker Haddock wrote:
 On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
  Hi there,
  
  I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO
  line.  The reason being is that Voicetronix sends out the DTMF too fast even
  before the line is fully established with the carrier.  Usually when dialing
  an 8 digit number, only 7 digits are actually successfully heard by the
  carrier.
  
  Currently, my dial plan is:
  exten = _9.,1,Dial(vpb/1-1/${EXTEN:1})
  
  Daniel said to insert a , before the numbers.  I am not too sure where to
  insert it.  I tried
  exten = _9.,1,Dial(vpb/1-1/,${EXTEN:1}) and that seems to be cause a
  parsing error.
  
  Anybody has any ideas for a hack?
  
  David
 The token to insert a pause is `W` (must be upper case).  Try this:
 
  exten = _9.,1,Dial(vpb/1-1/W${EXTEN:1})
 
 each `W` will cause a 0.5 second pause.
-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
Asterisk Resource Pages.

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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[Asterisk-Users] SIP Transfer problem

[Ariel's M-tech account]

I have been following and reading about the SIP 
problem of transferring calls with Asterisk. I did not seethis 
problem as havinga fix orhaving apatch for it. I can not 
use the # in our system due to IVR systems we access. 

Can someone let me know at what stage this is 
at. This is a major problem with our system in deploying SIP phones. 
We have Cisco 7960, Snom 200 and IpDialog's working but can not transfer. 


Thank you

Re: [Asterisk-Users] P2P RTP without SIP re-invites

[WipeOut]

Low, Adam wrote:

I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast.

So with that assumption I imagine a platform that would not get involved with the actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each end of the call deal with RTP encoding with their dedicated DSP hardware. There is an alternative in mind that maybe I could utilise some old Dialogic DSP cards that we have but I suspect trying to get these working with Asterisk would be a lot of programming work that I probably couldn't manage, maybe I'm wrong ?

The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd (specifically SIP breaks and calls are not torn down correctly) and of course you lose a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when it is in the SIP signalling path.

I vaguely remember previous discussions on this and even a patch but I am unable to find anything in the archives, does anybody have any info on that ?

The conclusion I have come to is that I would try and patch the Asterisk code. The idea being that when the RTP parameters are negotiated that Asterisk would pass through the source address/port from each SIP client causing them to talk RTP directly. I intend to begin work on this this weekend but am I hoping that maybe somebody else has already achieved what I desire, anybody ?

Rgds,
Adam


 

Asterisk single system scaling is an issue that I have been thinking 
about as well, and wondering about ways to cluster multiple Asterisk 
servers together to act as a unified system.. So far I haven't really 
got anywhere becasue everytjing I have thought of has been a problem 
most related to RTP..

Of course remember that the RTP is not really that much of a problem 
(apart from the bandwidth usage) when both the UA's are using the same 
codec.. Asterisk will simply switch the encoded voice traffic..

I am sure some clever person will come up with an answer but whether or 
not they share it is another question..

later..

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Re: [Asterisk-Users] PHP developer Wanted ! :-)

[NetOne Administrator]

What is your * app?
What should the frontend do?
Greetings,
Doichin Dokov
reseaux wrote:

Dear ALL
	i need to develop a web frontend for my * app i need only manage data from 
MySQL db, i will pay to develop it (not much :-) )
Thanks in advance
Dimitri

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Re: [Asterisk-Users] Calls dropping off

[Steve Foy]

Thanks, I'll try that and see how it goes.

Cheers,
Steve

On Fri, Jan 30, 2004 at 11:46:05AM -0500, Bill Hamel wrote:
 Try adding it to the phones involved so it looks like this:
 
 ; Shirley
 [100]
 type=friend
 username=xxx
 secret=xxx
 host=dynamic
 dtmfmode=rfc2833
 callerid=Shirley O'Neill 100
 context=internal
 [EMAIL PROTECTED]
 qualify=yes
 canreinvite=no
 
 -b
 
 
 Quoting Steve Foy [EMAIL PROTECTED]:
 
  Bill,
  
  On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote:
   Shot in the dark here ...
   
   Do you have: 
   
   canreinvite=no
   
   Set in sip.conf for the SIP phones in question ?
  
  No, I don't.
  
  All I have in sip.conf is the general stuff like:
  
 [general]
 port = 5060   ; Port to bind to (SIP is 5060)
 bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
  
 allow=all
 allow=GSM
 allow=G729
 allow=iLBC
 allow=SpeeX; Allow all codecs
 allow=ulaw
  
  and then about 10 friends like this:
  
 ; Shirley
 [100]
 type=friend
 username=xxx
 secret=xxx
 host=dynamic
 dtmfmode=rfc2833
 callerid=Shirley O'Neill 100
 context=internal
 [EMAIL PROTECTED]
 qualify=yes
  
  -- 
  Steve Foy|  http://www.unite.net
  UNITE Solutions  |  Tel: 028 9077 7338 
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  dangerous content by The CCIS.net MailScanner, and is
  believed to be clean.
  
  
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[Asterisk-Users] Compiling while * is running

[Stephen R. Besch]

I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my 
test  machine (ASUS A7V, 900 MHZ). However, If I try to compile on my 
production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the 
zaptel and asterisk compiles seg fault. I am assuming that they will 
compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 
compiled and is now running.

Is there a way to safely compile while * is running, so that I can 
minimize down time of the server?

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[Asterisk-Users] has Allison said this ?

[Lance Arbuckle]

Does anyone know if Allison has recorded anything along the lines of:

You don't have permission to dial that number.

Thanks.

--Lance Arbuckle
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Re: [Asterisk-Users] How to delay dialing

[Tilghman Lesher]

On Friday 30 January 2004 10:15, Walker Haddock wrote:
 On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
  Hi there,
 
  I am trying to delay sending out DTMF from Voicetronix OpenLine4
  to the CO line.  The reason being is that Voicetronix sends out
  the DTMF too fast even before the line is fully established with
  the carrier.  Usually when dialing an 8 digit number, only 7
  digits are actually successfully heard by the carrier.
 
  Currently, my dial plan is:
  exten = _9.,1,Dial(vpb/1-1/${EXTEN:1})
 
  Daniel said to insert a , before the numbers.  I am not too sure
  where to insert it.  I tried
  exten = _9.,1,Dial(vpb/1-1/,${EXTEN:1}) and that seems to be
  cause a parsing error.
 
  Anybody has any ideas for a hack?
 
  David

 The token to insert a pause is `W` (must be upper case).  Try this:

1)  The 'W' character is only for the zaptel channel.
2)  It's case insensitive (i.e. it does NOT need to be uppercase).
See line 2387 of zaptel.c if you'd like to confirm this for yourself.
3)  There is no current way within Asterisk to insert a pause into the
Voicetronix driver.
4)  There is no current way within Asterisk to insert a flash-hook
into the Voicetronix driver.
5)  The solution for 3 and 4 is attached.  This patch will allow you
to use the 'w' OR the 'W' character to insert a pause and to use the
'f' or 'F' character to insert a flash-hook.  Please note (VERY
IMPORTANT): in the Voicetronix driver, the pause is 1.0 seconds, not
0.5 seconds, like it is in the Zaptel driver.

-Tilghman
Index: channels/chan_vpb.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_vpb.c,v
retrieving revision 1.12
diff -u -r1.12 chan_vpb.c
--- channels/chan_vpb.c 9 Dec 2003 23:55:17 -   1.12
+++ channels/chan_vpb.c 30 Jan 2004 17:27:28 -
@@ -681,13 +681,26 @@
 {
 struct vpb_pvt *p = (struct vpb_pvt *)ast-pvt-pvt;
 int res = 0;
-char *s = strrchr(dest, '/');
+char *s = strrchr(dest, '/'), char *t;
 
 if (s)
 s = s + 1;
 else
 s = dest;
 
+   /* We cannot use either the  or the , in a Dial string, as these
+* characters are used to signal 1) different concurrent technologies,
+* or 2) separation of application arguments.  Therefore, this channel
+* driver should translate the w (for a pause) to the , and the f (for
+* a flash-hook) to a . */
+
+   for (t = s; t != '\0' ; t++) {
+   if ((*t == 'w') || (*t == 'W'))
+   *t = ',';
+   else if ((*t == 'f') || (*t == 'F'))
+   *t = '';
+   }
+
 if (ast-_state != AST_STATE_DOWN  ast-_state != AST_STATE_RESERVED) {
ast_log(LOG_WARNING, vpb_call on %s neither down nor reserved!\n, 
ast-name);

RE: [Asterisk-Users] IAX call problems

[Steven Sokol]

Rattana,

I have had the same problem with IAX Phone.  I think there is still
something slightly off in iaxClient_lib.c or one of the associated files.  I
am trying to figure it out myself.  Please send me any additional debugging
files as you generate them.

Thanks,

Steve


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rattana BIV
Sent: Friday, January 30, 2004 7:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX call problems

hi,
 
I use IAX softphone with asterisk and I notice that a call between two IAX
softphones end after 1 min. Then I can't hear anything but the call still in
progress.
I have this log in asterisk IAX debug:
 
Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
  Timestamp: 00016ms  SCall: 21589  DCall: 1
[192.168.1.22:4569]   
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE   Subclass: 2
   Timestamp: 65795ms  SCall: 6  DCall: 21588
[192.168.1.22:4569]   
Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE   Subclass: 2
   Timestamp: 65795ms  SCall: 6  DCall: 21588
[192.168.1.22:4569]   
Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK
  Timestamp: 65795ms  SCall: 21588  DCall: 6
[192.168.1.22:4569]   
Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: PING
 Timestamp: 75906ms  SCall: 22105  DCall: 5
[192.168.1.77:4569]   
Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK
 Timestamp: 75906ms  SCall: 5  DCall: 22105 [192.168.1.77:4569]
 
 
Any suggestions ???
 
 
Thanks in advance
 
Rattana
 
 
PS: The softphone I use work with wiax.dll and is developpe by me =)
 
 
 
 
 
 


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Re: [Asterisk-Users] has Allison said this ?

[WipeOut]

Lance Arbuckle wrote:

Does anyone know if Allison has recorded anything along the lines of:

You don't have permission to dial that number.

 

Or a more versitile way of saying it..

The number you dialed is not permitted.

This could then mean that *you* are not allowed to dial it or the 
*system* does not allow that number.. :)

later..

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Re: [Asterisk-Users] has Allison said this ?

[David Gomillion]

Lance Arbuckle wrote:
 Does anyone know if Allison has recorded anything along the lines of:
 
 You don't have permission to dial that number.

I think so... under tt-monkeys.gsm.

 
 Thanks.

You're welcome.


PS. Sorry, I couldn't resist on this one.

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Re: [Asterisk-Users] HANGUPCAUSE

[Eric Wieling]

See Bug Number 890 on bugs.digium.com.

--Eric

 From: Tais M. Hansen [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, January 30, 2004 9:20 AM
 Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
 
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 On Friday 30 January 2004 00:57, Eric Wieling wrote:
  Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
  being translated into Asterisk Cause Codes and being passed back to
  app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
  variable)?
 
 HANGUPCAUSE is working fine here (cvs).


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RE: [Asterisk-Users] List traffic

[Dawid Mielnik]

Michael,

I have the same thing -- 1 to 4 posts a day !

regards,

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael Graves
Sent: Thursday, January 29, 2004 12:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] List traffic


All of a sudden my list traffic appears to have dropped to a few
messages/day the past few days. I anyone else seeing this as well?

Michael


--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

...I believe in love, its all we've got. - Elton John

** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704


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[Asterisk-Users] MeetMe Video option

[Regovich, Timothy]

Hello All:

Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.

Is there something else that I need to be doing other than set the v flag
on my extension for the meetme app?

Thanks,

Tim


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Re: [Asterisk-Users] Compiling while * is running

[Rob Fugina]

On Fri, Jan 30, 2004 at 12:21:49PM -0500, Stephen R. Besch wrote:
 I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my 
 test  machine (ASUS A7V, 900 MHZ). However, If I try to compile on my 
 production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the 
 zaptel and asterisk compiles seg fault. I am assuming that they will 
 compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 
 compiled and is now running.
 
 Is there a way to safely compile while * is running, so that I can 
 minimize down time of the server?

Seg faulting compiles usually indicate a memory problem on the machine.
Not lack of size, but bad memory, badly seated memory, etc...  There's
no reason asterisk running, or the drivers being loaded, should
cause a compile to seg fault.

On the other hand, the load of a compile could affect asterisk's
performance...

Rob

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My firewall filters MS Office attachments.

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RE: [Asterisk-Users] Adtran 750 DID question.

[Bisker, Scott (7805)]

I tried both featd and em in zapata.conf, to no avail.  I restarted in between all 
changes.  Is it possible to signal the DPO ports on the 750 with fxo_ls or fxo_ks?

This is the last piece to my DID puzzle.  Anyone else with experience on this oddball 
config?

Thanks,

-sb

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Sharp
Sent: Friday, January 30, 2004 11:52 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Adtran 750 DID question.


 Hello All,

 I've mostly solved my DID problem from a few days ago.  Apparenly the
 lines weren't configured properly.  Now heres the next question.  12 EM
 wink lines from telco.  I have them all plugging into an Adtran 750 with
 FXS cards.  The Adtran ports are configured DPO.   How do I signal this
 from Zaptel.  I have them setup EM in zaptel.conf and EM_W in
 zapata.conf.  They work, however, no DNIS info is being passed.  Do I need
 to signal these something different like loopstart or kewlstart, so the
 DNIS info gets passed?  I watch the Tx/Rx bits from zttool, and everything
 looks okay coming from the Adtran.  It looks like asterisk isn't winking
 properly.

I had a similar problem.  I ended up setting the trunks to either just
plain em or featd (I don't remember).  I chased through the chan_zap
source code and decided (maybe incorrectly) that asterisk doesn't look for
DNIS digits in EM Wink mode.
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Re: [Asterisk-Users] How to delay dialing

[Walker Haddock]

On Fri, Jan 30, 2004 at 10:58:15AM -0600, Eric Wieling wrote:
 Are you sure this works for VoiceTronix Driver?  It's not implemented in
 app_dial, but in chan_zap.
 
I've only used it with a zap device.  Sorry I didn't think this through.


 On Fri, 2004-01-30 at 10:15, Walker Haddock wrote:
  On Thu, Jan 29, 2004 at 04:10:42PM -0800, David Liu wrote:
   Hi there,
   
   I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO
   line.  The reason being is that Voicetronix sends out the DTMF too fast even
   before the line is fully established with the carrier.  Usually when dialing
   an 8 digit number, only 7 digits are actually successfully heard by the
   carrier.
   
   Currently, my dial plan is:
   exten = _9.,1,Dial(vpb/1-1/${EXTEN:1})
   
   Daniel said to insert a , before the numbers.  I am not too sure where to
   insert it.  I tried
   exten = _9.,1,Dial(vpb/1-1/,${EXTEN:1}) and that seems to be cause a
   parsing error.
   
   Anybody has any ideas for a hack?
   
   David
  The token to insert a pause is `W` (must be upper case).  Try this:
  
   exten = _9.,1,Dial(vpb/1-1/W${EXTEN:1})
  
  each `W` will cause a 0.5 second pause.
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Re: [Asterisk-Users] MeetMe Video option

[WipeOut]

Regovich, Timothy wrote:

Hello All:

Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
 

What video phone did you use?

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RE: [Asterisk-Users] Auto dial in Off Hook situation.

[David J Carter]

Thanks John,


I think it is not that simple. I am not using a phone but a Cisco ATA.

The scenario: -

User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension

The Multitech MVP100 used to connect to my old analogue switch which was set
to auto call one extension.

The old switch died, (rest it's soul), and I have built the * to replace,
(nay superseded) it. Lot more functions
for less of the greenbacks.

So it is really the Cisco ATA that I need to auto call an extension.

Just to cap it all I can't seem to get into the web interface of the Cisco
at present, Keep getting Invalid Access.


regards


Dave
SipPhone: - 1-747-386-2964
IaxTel: - 1-700-818-8820

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: 30 January 2004 14:23
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto dial in Off Hook situation.


Hi all,


I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.


Anyone done this or know how it's done.


regards

Dave

Depends on the phone.  If you have an FXS interface, look for
immediate= in your zapata.conf file.

If you have an IP phone, search the vendor's documentation for PLAR
(Private Line Auto Ringdown)

JT
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RE: [Asterisk-Users] MeetMe Video option

[Regovich, Timothy]

I have written my own.  Java(JMF) based.
It is pretty rudimentary, but does handle audio (gsm, ulaw) and video (jpeg
and H263).

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, January 30, 2004 1:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] MeetMe Video option


Regovich, Timothy wrote:

Hello All:

Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.

  

What video phone did you use?

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RE: [Asterisk-Users] MeetMe Video option

[Jonathan Moore]

I would also be interested in this in regards to working with D-Link
videophones. They use the same setup as netmeeting h.263, but with another rfc
add on. I know current OpenH323 configs do not quite work with it, but I saw a
post that it is in cvs working using a patch to ffmpeg.


-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Regovich, Timothy [EMAIL PROTECTED]:

 I have written my own.  Java(JMF) based.
 It is pretty rudimentary, but does handle audio (gsm, ulaw) and video (jpeg
 and H263).
 
 Tim
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
 Sent: Friday, January 30, 2004 1:30 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] MeetMe Video option
 
 
 Regovich, Timothy wrote:
 
 Hello All:
 
 Has anyone configured a meetme conference to use video?
 I have successfully used video phones to talk through *, but I cannot seem
 to get video when those phones dial into a meetme conference.
 
   
 
 What video phone did you use?
 
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 Notice:  This e-mail message, together with any attachments, contains
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 Jersey, USA 08889), and/or its affiliates (which may be known outside the
 United States as Merck Frosst, Merck Sharp  Dohme or MSD and in Japan, as
 Banyu) that may be confidential, proprietary copyrighted and/or legally
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 named on this message.  If you are not the intended recipient, and have
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Visit Winfield Public Schools at http://usd465.com
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[Asterisk-Users] Newbridge Mainstreet 3624

[David_Cox]

I've got a Newbridge CB hanging on the wall not being used right now and
I'd like to hear opinions on using it with Asterisk. If anyone has a manual
for it I'd like to get a copy of it. I tried the googling approach but
turned up nothing much except a Tech manual if I want to change out control
boards.

Thanks

David Cox
Director of Information Technology
Ramtex, Inc.
http://www.ramtex.com


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RE: [Asterisk-Users] Auto dial in Off Hook situation.

[James Sharp]

 Thanks John,


 I think it is not that simple. I am not using a phone but a Cisco ATA.

 The scenario: -

 User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
 (FXO))--Cisco ATA--Asterisk--Any extension

Any reason you can't use the H.323 load for the MVP200?  I've not tried it
in a year or so, but it mostly worked last time I tried it.
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Re: [Asterisk-Users] Adtran 750 DID question.

[Michael Welter]

Did you say you were using Adtran FXS cards?

Bisker, Scott (7805) wrote:

Hello All,

I've mostly solved my DID problem from a few days ago.  Apparenly the lines weren't configured properly.  Now heres the next question.  12 EM wink lines from telco.  I have them all plugging into an Adtran 750 with FXS cards.  The Adtran ports are configured DPO.   How do I signal this from Zaptel.  I have them setup EM in zaptel.conf and EM_W in zapata.conf.  They work, however, no DNIS info is being passed.  Do I need to signal these something different like loopstart or kewlstart, so the DNIS info gets passed?  I watch the Tx/Rx bits from zttool, and everything looks okay coming from the Adtran.  It looks like asterisk isn't winking properly.

When I had the lines misconfigured for fxs_ls the DNIS info was passing fine.

I'm running 
zaptel-0.8.0
libpri-0.5.1
And asterisk CVS from 12/23/2003
RedHat 8.0
Dual 2.4 Xeon Processors (hyperthreading disabled)
2Gig Memory

Any help would be greatly appreciated.

Regards,

-sb
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RE: [Asterisk-Users] MeetMe Video option

[Regovich, Timothy]

I was wondering if it was supported, and how.

It seems to me that video conferencing is a different beast than audio
conferencing because you cannot simply mix video like you can mix audio.

The conferencing server would have to 
1) mix the video by creating one aggregate outbound paneled type window,
or
2) have each incoming stream sent to each registered listening stream, which
is ok, as long as the client can handle multiple incoming streams reasonably
(yes, I realize that this results in n*n bandwidth usage), or 
3) the conference serer would need to designate a master video stream and
ignore all other incoming streams.

Each of these seem to be viable options, depending on what you want to do.

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp
Sent: Friday, January 30, 2004 1:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] MeetMe Video option


Citeren Regovich, Timothy [EMAIL PROTECTED]:

 Has anyone configured a meetme conference to use video?
 I have successfully used video phones to talk through *, but I cannot seem
 to get video when those phones dial into a meetme conference.

Cool, what devices are you using ? Would love to try some :-)

 Is there something else that I need to be doing other than set the v
flag
 on my extension for the meetme app?

Hmm, don't think that's supported yet ??


-- 
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
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RE: [Asterisk-Users] Auto dial in Off Hook situation.

[John Todd]

Yes, it is that simple, but of course there is a precursor 
requirement that you need to 1) be able to configure  your ATA, and 
2) know how to configure your ATA.

Google is your friend.  Please use Google before you reply back with 
additional questions; it saves us all time and email bandwidth.

http://www.google.com/search?hl=enie=UTF-8oe=UTF-8q=%22ata-186%22+plarbtnG=Google+Search

I'm somewhat confused as to why you would ever put those multitechs 
in there when you could go directly from the ATA to Asterisk, but I'm 
assuming you have some good reason.

JT

At 6:30 PM + 1/30/04, David J Carter wrote:
Thanks John,

I think it is not that simple. I am not using a phone but a Cisco ATA.

The scenario: -

User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
(FXO))--Cisco ATA--Asterisk--Any extension
The Multitech MVP100 used to connect to my old analogue switch which was set
to auto call one extension.
The old switch died, (rest it's soul), and I have built the * to replace,
(nay superseded) it. Lot more functions
for less of the greenbacks.
So it is really the Cisco ATA that I need to auto call an extension.

Just to cap it all I can't seem to get into the web interface of the Cisco
at present, Keep getting Invalid Access.
regards

Dave
SipPhone: - 1-747-386-2964
IaxTel: - 1-700-818-8820
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: 30 January 2004 14:23
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto dial in Off Hook situation.

Hi all,

I have looked through the wiki for any information on how to make an
extension autodial another extension when it goes off hook.
Anyone done this or know how it's done.

regards

Dave
Depends on the phone.  If you have an FXS interface, look for
immediate= in your zapata.conf file.
If you have an IP phone, search the vendor's documentation for PLAR
(Private Line Auto Ringdown)
JT
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RE: [Asterisk-Users] Adtran 750 DID question.

[Bisker, Scott (7805)]

Yes.  Adtran FXS cards.




Did you say you were using Adtran FXS cards?

Bisker, Scott (7805) wrote:

 Hello All,
 
 I've mostly solved my DID problem from a few days ago.  Apparenly the lines weren't 
 configured properly.  Now heres the next question.  12 EM wink lines from telco.  I 
 have them all plugging into an Adtran 750 with FXS cards.  The Adtran ports are 
 configured DPO.   How do I signal this from Zaptel.  I have them setup EM in 
 zaptel.conf and EM_W in zapata.conf.  They work, however, no DNIS info is being 
 passed.  Do I need to signal these something different like loopstart or kewlstart, 
 so the DNIS info gets passed?  I watch the Tx/Rx bits from zttool, and everything 
 looks okay coming from the Adtran.  It looks like asterisk isn't winking properly.
 
 When I had the lines misconfigured for fxs_ls the DNIS info was passing fine.
 
 I'm running 
 zaptel-0.8.0
 libpri-0.5.1
 And asterisk CVS from 12/23/2003
 RedHat 8.0
 Dual 2.4 Xeon Processors (hyperthreading disabled)
 2Gig Memory
 
 Any help would be greatly appreciated.
 
 Regards,
 
 -sb
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Re: [Asterisk-Users] IAX call problems

[Dan Tucny]

Hi Rattana,

Do you have jitterbuffer enabled?

Dan

On Fri, 2004-01-30 at 13:40, Rattana BIV wrote:
 hi,
  
 I use IAX softphone with asterisk and I notice that a call between two
 IAX softphones end after 1 min. Then I can't hear anything but the
 call still in progress.
 I have this log in asterisk IAX debug:
  
 Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass:
 ACK
   Timestamp: 00016ms  SCall: 21589  DCall: 1
 [192.168.1.22:4569]   
 Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE   Subclass:
 2
Timestamp: 65795ms  SCall: 6  DCall: 21588
 [192.168.1.22:4569]   
 Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE   Subclass:
 2
Timestamp: 65795ms  SCall: 6  DCall: 21588
 [192.168.1.22:4569]   
 Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass:
 ACK
   Timestamp: 65795ms  SCall: 21588  DCall: 6
 [192.168.1.22:4569]   
 Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass:
 PING
  Timestamp: 75906ms  SCall: 22105  DCall: 5
 [192.168.1.77:4569]   
 Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass:
 ACK
  Timestamp: 75906ms  SCall: 5  DCall: 22105
 [192.168.1.77:4569]
  
  
 Any suggestions ???
  
  
 Thanks in advance
  
 Rattana
  
  
 PS: The softphone I use work with wiax.dll and is developpe by me =)
  
  
  
  
  
  

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Re: [Asterisk-Users] Compiling while * is running

[David Gomillion]

Rob Fugina wrote:
[snip]
 Is there a way to safely compile while * is running, so that I can
 minimize down time of the server?

 Seg faulting compiles usually indicate a memory problem on the
 machine. Not lack of size, but bad memory, badly seated memory,
 etc...  There's no reason asterisk running, or the drivers being
 loaded, should
 cause a compile to seg fault.

I don't agree.  When first learning to program, my programs segfaulted all
of the time, regarless of what machine I was on.  Often, it was doing
something stupid, like trying to replace a file that was in use, etc.

On my machine, compiling took ~2 minutes, for all 3 pieces (zaptel, libpri,
and asterisk).  To get 5 9's (99.999% uptime), you need to be up for 13.9
days (check my math... it's been a while).

My suggestion: if this downtime is unacceptable for your use, then get an
identical machine, exactly alike in all ways, including library versions,
hardware, etc, and compile it on that machine.  Then copy the appropriate
directories over to your production machine.  Copy the production machine's
directories to a safe location, stop * and zaptel, copy the new compiled
things over, then restart * and zaptel.  My guess is that 30 seconds should
be plenty of time for this change.  Thus, you only need to have been up for
the last 3.47 days to have 99.999% uptime.

Either that, or maybe if uptime is so critical, you should have a hot
spare machine on-hand at all times.

Anyway, just some thoughts.

David Gomillion

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RE: [Asterisk-Users] G729 license

[Wes Marderness]

I 
purchased a license from Digium, If you ask they will can also give you a trial 
license to test out.

Wes

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Jess 
  MagnayeSent: Friday, January 30, 2004 10:29 AMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] G729 
  licenseImportance: High
  Hello all,
  
  I would like to just verify where to purchase the 
  G729 license for Asterisk. Like I want to run G729 codec for all my 
  calls passing thru Asterisk (voicemail, parking, via ZAP, via SIP, etc). 
  The list says license is taken from Digium, does that apply also if I have 
  Dialogic cards on my *?
  
  

Re: [Asterisk-Users] Compiling while * is running

[Joe Phillips]

On Fri, 2004-01-30 at 14:26, David Gomillion wrote:
 Rob Fugina wrote:

  Seg faulting compiles usually indicate a memory problem on the
  machine. Not lack of size, but bad memory, badly seated memory,
  etc...  There's no reason asterisk running, or the drivers being
  loaded, should
  cause a compile to seg fault.
 
 I don't agree.  When first learning to program, my programs segfaulted all
 of the time, regarless of what machine I was on.  Often, it was doing
 something stupid, like trying to replace a file that was in use, etc.

I think you are mis-reading Rob.  True that your own programs segfaulted
but did you cause GCC to segfault?  I think the original author said
that GCC was itself segfaulting.  GCC is so well used and tested that as
Rob points out, the most common cause of a GCC segfault is hardware
failure.

 My suggestion: if this downtime is unacceptable for your use, then get an
 identical machine, exactly alike in all ways, including library versions,
 hardware, etc, and compile it on that machine.  Then copy the appropriate
 directories over to your production machine.  Copy the production machine's
 directories to a safe location, stop * and zaptel, copy the new compiled
 things over, then restart * and zaptel.  My guess is that 30 seconds should
 be plenty of time for this change.  Thus, you only need to have been up for
 the last 3.47 days to have 99.999% uptime.

This is a reason I argue for binary packages in production
environments.  You can build the packages (eg. debs or RPMs) on a
development machine at your leisure and install the binary in minutes on
the production machine.  If your packages use proper dependencies you
can also be much more sure you can reproduce your environment on new
hardware (testing, qa, hot-spare, disaster recovery etc).

-joe
-- 
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   Custom Internet and Computer Solutions
   Linux, UNIX, Java Training

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Re: [Asterisk-Users] Compiling while * is running

[Steven Critchfield]

On Fri, 2004-01-30 at 13:26, David Gomillion wrote:
 Rob Fugina wrote:
 [snip]
  Is there a way to safely compile while * is running, so that I can
  minimize down time of the server?
 
  Seg faulting compiles usually indicate a memory problem on the
  machine. Not lack of size, but bad memory, badly seated memory,
  etc...  There's no reason asterisk running, or the drivers being
  loaded, should
  cause a compile to seg fault.
 
 I don't agree.  When first learning to program, my programs segfaulted all
 of the time, regarless of what machine I was on.  Often, it was doing
 something stupid, like trying to replace a file that was in use, etc.

You apparently still have quite a bit more to learn. If you read the
first line quoted, you will see that it is the compiling that is a
problem. At no time during compile is the application you are compiling
actually executed. Only gcc and it's helpers should be executed. Gcc is
notorious for finding bad memory as it sprawls out over large sections
and is sensitive to bits flipping around. If Asterisk was segfaulting,
then there may be a question as to whether asterisk behaved differently
under load(timing issues) or if it was still bad memory.

 On my machine, compiling took ~2 minutes, for all 3 pieces (zaptel, libpri,
 and asterisk).  To get 5 9's (99.999% uptime), you need to be up for 13.9
 days (check my math... it's been a while).

5 9's is approximately 5 minutes over the course of a year. You couldn't
do this 3 times a year and stay under that time so that is every 4+
months. Also that is assuming that the modules unload and load fine, and
you aren't dealing with any problems getting sync back on any T1 lines.
Really any reload of the modules will put you close to that 5 minutes
per year. Luckily the low level drivers don't change often, and neither
does libpri. So updating and restarting asterisk usually only incurs a
sub 1 minute unavailable period.

 My suggestion: if this downtime is unacceptable for your use, then get an
 identical machine, exactly alike in all ways, including library versions,
 hardware, etc, and compile it on that machine.  Then copy the appropriate
 directories over to your production machine.  Copy the production machine's
 directories to a safe location, stop * and zaptel, copy the new compiled
 things over, then restart * and zaptel.  My guess is that 30 seconds should
 be plenty of time for this change.  Thus, you only need to have been up for
 the last 3.47 days to have 99.999% uptime.

You should really look into bc -l before you speak. 30 seconds over 3.47
days is 99.989 percent uptime. For true 5 9's, you could only spare
2.998 seconds in 3.47 days.

 Either that, or maybe if uptime is so critical, you should have a hot
 spare machine on-hand at all times.

Maybe you don't know how long it takes to sync a T1 line. That alone
_can_ take almost a minute. Then the service can come up. If time is
critical, it is probably not a good idea to just upgrade asterisk at a
whim. This is why a previous post to dev by myself showed I'm still
running releases from October and November of last year. Nothing in the
newer releases are needed at this time, and therefore upgrading isn't
important.
-- 
Steven Critchfield  [EMAIL PROTECTED]

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[Asterisk-Users] error on IAX1.conf and warning on chan_iax2.c

[Michael Zheng]

Hi, 

I have a wildcard x100p. I just installed asterisk by
following step:

# cd ../zaptel
# make clean ; make install
# cd ../libpri
# make clean ; make install
# cd ../asterisk
# make clean ; make install
# make samples 

When I test Asterisk typing
# asterisk –c

I find one error and one warning:

 [chan_iax.so] = (Inter Asterisk eXchange)
  == Manager registered action IAX1peers
  == Parsing '/etc/asterisk/iax1.conf': Not found (No
such file or directory)
Jan 30 14:38:23 ERROR[1074468608]: chan_iax.c:4826
set_config: Unable to load config iax1.conf
  == Parsing '/etc/asterisk/iax.conf': Found
  == Using TOS bits 16
  == Registered channel type 'IAX1' (Inter Asterisk
eXchange Drver)
  == Registered channel type 'IAX' (Inter Asterisk
eXchange Drver)
  == IAX Ready and Listening on 0.0.0.0 port 5036

 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
Jan 30 14:38:24 WARNING[1074468608]: chan_iax2.c:5510
set_config: Ignoring port for now
  == Registered channel type 'IAX2' (Inter Asterisk
eXchange Driver (Ver 2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569

What is IAX1.conf, what I don't have this file? Why I
get warning on chan_ian2.c? How can I solve these
problems?

By the way, if I only use wildcards (wildcard x100p
and wildcard TDM400p), when I install Asterisk, can I
skip installing libpri just do like this:
# cd ../zaptel
# make clean ; make install
# cd ../asterisk
# make clean ; make install
# make samples 
(skip: cd ../libpri; make clean; make install)

Best,
Michael

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[Asterisk-Users] X-Lite, X100P, and Speex

[Kostur, Andre]

Title: X-Lite, X100P, and Speex





I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec. I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works. However, if I call out the analog port it appears that audio being initiated by X-Lite is being dropped, but audio being initiated from the analog line is being encoded and heard OK on X-Lite. /var/log/asterisk/messages keeps repeating WARNING[]: Frame too large and WARNING[]: Out of buffer space over and over again. Any ideas on what's wrong? (and if it's simply that one cannot use the speex codec with outbound calls, how would one configure asterisk to allow speex when it's a SIP to SIP call, but G.711 if it's a SIP to Analog call?)

Oh, and using ztmonitor, it shows the zap channel receiving all sorts of sound, but no transmit.


Asterisk 0.7.1 (Debian/Unstable package)
zaptel 0.1.6

[Asterisk-Users] Extension Questions

[Shad Mortazavi]

Dear all,



I have the following lines in my extentions.conf file;



;All US Calls

exten = _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) 

;Dial 9 for outgoing numbers

exten =_9.,1,Dial(Zap/g1/${EXTEN:1})



;include Brunswick

switch = IAX2/dornoch:[EMAIL PROTECTED]/sip



What Im trying to do is to send any calls starting
with 9001 out through my system in the USA and any number starting with a
9 through my local number.



However what ever the number I dial starting with a 9 goes
out of the local interface. If I comment out the exten
=_9.,1,Dial(Zap/g1/${EXTEN:1}) then it works.



How can I make the line starting exten = _9001 take precedence
over the line starting exten = _9001?



Kind Regards



Shad Mortazavi

-

US Technical Manager

Nexus Management

[Asterisk-Users] Re: MeetMe Video option

[Matt Lawson]

That's one of the things that's been on our (1control, I have nothing to 
do with Digium) wishlist/to do list that just hasn't gotten done yet.

Currently, video in meetme is not supported.  What we experience is the 
audio will conference with the other audio streams but the video just 
freezes.  I was hoping to look into someday but I'm swamped with 1000 
other things of higher priority.  I have been thinking though, of some 
ways it could be supported, starting with the simplest and easiest:

1.  First, if only 2 of the phones in the conference are video phones, 
allow them to exchange their video with each other, while having all of 
the audio streams conferenced as usual.

2a.  The next step could be having each videophone rotate which stream 
it was showing for a few seconds (20 seconds maybe?).  i.e. you could 
have 3 video calls mixed with several audio-only calls.  Initially video 
call #1 would show #2's image, #2 would show #3's image, #3 would show 
#1's image for a few seconds, then rotate them by 1.  Of course you 
don't need to show your own!  :)  Actually, ours has a 
picture-in-picutre in the corner so you can see yourself all the time 
anyway.

2b.  The other option instead of time-rotating the images would be to 
try to show the image of whoever was talking.  That kind of sounds like 
a pain to me, but maybe it's doable.

3.  The really fancy thing would be to have Asterisk decode all of the 
video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them 
and send them to each client.  That REALLY sounds like a pain to me, but 
again, maybe it's doable.

Right now I'd be pretty happy with 2a though.

- Matt



Message: 3
From: Regovich, Timothy [EMAIL PROTECTED]
To: '[EMAIL PROTECTED]' [EMAIL PROTECTED]
Date: Fri, 30 Jan 2004 13:07:46 -0500
Subject: [Asterisk-Users] MeetMe Video option
Reply-To: [EMAIL PROTECTED]
Hello All:

Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
Is there something else that I need to be doing other than set the v flag
on my extension for the meetme app?
Thanks,

Tim
 



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Re: [Asterisk-Users] Extension Questions

[Steven Critchfield]

On Fri, 2004-01-30 at 14:00, Shad Mortazavi wrote:
 Dear all,
 
  
 
 I have the following lines in my extentions.conf file;
 
  
 ;All US Calls
 
 exten =
 _9001XX,1,Dial(IAX2/dornoch:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) 
   
 
 ;Dial 9 for outgoing numbers
 
 exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
 
  
 ;include Brunswick
 
 switch = IAX2/dornoch:[EMAIL PROTECTED]/sip
 
  
 What Im trying to do is to send any calls starting with 9001 out
 through my system in the USA and any number starting with a 9 through
 my local number.
 
  
 
 However what ever the number I dial starting with a 9 goes out of the
 local interface. If I comment out the exten
 =_9.,1,Dial(Zap/g1/${EXTEN:1}) then it works.
 
  
 
 How can I make the line starting exten = _9001 take precedence over
 the line starting exten = _9001?

Well, you could define your _9. a little more clearly. Is it likely that
your other calls would have some digit other than a 0 following the 9?
If so then you should see the need to use wither a a N which is 2-9 or
maybe some other class in the regex. 
-- 
Steven Critchfield  [EMAIL PROTECTED]

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