Re: [asterisk-users] NTL Hangup

[Kyle Gordon]

Leo Ann Boon wrote:

Kyle Gordon wrote:

Hi all,

I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P 
cheapo card.


The problem lies with detecting when the far end has hung up. It fails 
to detect it, and will only cleardown when the silence timeout has 
been reached. Now, I've seen the thread at 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg32337.html, 
to which nothing has come of it. That was almost 2 years ago, so I was 
wondering if there's been any progress?

2 things:
a. You need to show us your zaptel.conf and zapata.conf.
b. Do you know the tone plan used by ntl? I guess it should be the UK 
standard.


Leo


Good point :-)

=== zaptel.conf ===

fxsks=1 #X100P
defaultzone=uk
loadzone=uk

=== zapata.conf ===

[channels]
; general settings
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
immediate=no
callprogress=no
busydetect=yes

; to enable/disable music onhold
musiconhold=default

; This tells asterisk to try and kill line
; echo using software detection.
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes

switchtype=national
signalling=fxs_ks
context=ntl_pstn
channel=1 ;X100P

I don't know the tone plan for NTL. They seem to use a different tone 
for hanging up from BT, but I'm not sure how to go about implementing 
any changes in the configs to reflect it.


Regards

Kyle
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RE: [asterisk-users] TE110P and HDLC problems

[Lee Archer]

I had this problem and in the end it appeared to be slot timing on the mobo.  I 
had to put the TE110P in the 1st slot - which happened to be a PCI-X slot.  
That was using a Supermicro motherboard too. 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew 
Fredrickson
Sent: 25 January 2007 20:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE110P and HDLC problems

There was a recent driver fix that *might* help you.  It's not in an official 
1.x.x release yet, but if you check out 1.2 from svn, you should get the latest 
version of the driver with the fix.

Matthew Fredrickson

On Jan 25, 2007, at 9:15 AM, Marc Patino Gómez wrote:

 Hi!,

 this issue makes me crazy. I read a lot of docs, also * mailling list 
 and I try a lot of things  without success.

 Any help will be appreciated. Here is the info:

 Hardware:
 
 Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 
 5050 Digium TE110P

 Software
 -
 Asterisk version 1.2.12.1
 Zaptel version 1.2.8

 /etc/zaptel.conf

 loadzone=es
 defaultzone=es
 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15
 dchan=16
 bchan=17-31

 The dammed errors:

 Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
 event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 25 14:11:52 
 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
 event: HDLC Abort (6) on Primary D-channel of span 1 Jan 25 14:11:52 
 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got
 event: HDLC Abort (6) on Primary D-channel of span 1 ...

 I tried the following without success:

 - Disable Hyper Threading.
 - Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, 
 so TE110P has his own IRQ as shows lspci -vb.
 - Also I tried with APIC and without APIC.
 ..


 These HDLC errors appear when I physically loop the E1 interface in 
 the Card and also appear, and more frequently, when I connect the E1 
 circuit (from the Telco) to the interface of the Card.


 Thanks a lot

 --  
 --- 
 -

 Marc Patino Gómez
 Dpto. Sistemas

 Claranet España. Servicios Internet
 C/General Almirante 2-28, Torres Cerdá
 08014 Barcelona
 Tel. Información General: 902 884 633 Tel. Soporte Técnico: 902 884 622
 Fax: +34 93 445 19 20
 www.claranet.es

 Claranet Group: United Kingdom - Spain - France - Germany - Portugal -  
 Netherlands - USA

 --- 
 -

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[asterisk-users] Sangoma card dying after 1hour

[Jon Schøpzinsky]

Hello List

 

I am having a rather big problem with a sangoma A104 card, I just installed to 
replace a Digium TE410 card, that was acting up.

 

But now we have a problem with the sangoma card. It runs great after being 
started, and calls proceed as normal, but after about 1 hour, it stops being 
able to make and receive calls.

If I run wanpipemon debug,  can see that the card still receives packets from 
the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk 
just responds with a:

NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - 
Circuit/channel congestion)

 

I am pretty shure that this is a configuration issue, but are there anything I 
need to be aware of when moving from a Digium card to a sangoma card?

 

Kind Regards
Jon Leren Schøpzinsky

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[asterisk-users] Re: How to exit from console?

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 E.g: because you have a valid PID file of the controlling process. If
 you actually want to kill it, you can.
 
 And you don't need physical access to the system to get to the one and
 only real console. OTOH, if you do have physical access, you have full
 control of Asterisk, as you may inject custom dialplan.

And if, for some reason Asterisk dies, you have to start it manually?


-- 
Tomislav Parcina
[EMAIL PROTECTED]
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Re: [asterisk-users] Sangoma card dying after 1hour

[Steve Davies]

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:


Hello List

I am having a rather big problem with a sangoma A104 card, I just installed
to replace a Digium TE410 card, that was acting up.

But now we have a problem with the sangoma card. It runs great after being
started, and calls proceed as normal, but after about 1 hour, it stops being
able to make and receive calls.

If I run wanpipemon debug,  can see that the card still receives packets
from the ISDN, but when I make a call, I cant see it in wanpipemon, and
asterisk just responds with a:

NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 -
Circuit/channel congestion)

I am pretty shure that this is a configuration issue, but are there anything
I need to be aware of when moving from a Digium card to a sangoma card?


Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded
as there are some resource leak fixes in that version.

Regards,
Steve
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[asterisk-users] pickup internal and external calls

[René Enskat]

hello,

i want to make a dialplan where i can pickup calls to an extension when
there are internal and external calls.
i want to use only one prefix for pickup both situations so there is a
plan how to check if the incoming call is an internal call or an
extern???


regards rene


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RE: [asterisk-users] Sangoma card dying after 1hour

[Jon Schøpzinsky]

I am running the newest version, from the sangoma wiki.

Jon

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma card dying after 1hour

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:

 Hello List

 I am having a rather big problem with a sangoma A104 card, I just installed
 to replace a Digium TE410 card, that was acting up.

 But now we have a problem with the sangoma card. It runs great after being
 started, and calls proceed as normal, but after about 1 hour, it stops being
 able to make and receive calls.

 If I run wanpipemon debug,  can see that the card still receives packets
 from the ISDN, but when I make a call, I cant see it in wanpipemon, and
 asterisk just responds with a:

 NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 -
 Circuit/channel congestion)

 I am pretty shure that this is a configuration issue, but are there anything
 I need to be aware of when moving from a Digium card to a sangoma card?

Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded
as there are some resource leak fixes in that version.

Regards,
Steve
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[asterisk-users] WellTech 380x Gateway

[Mark Coccimiglio]

Ok this is a simple question...

What has been your experience with the WellTech 38xx series (I'm looking 
specifically at the 3802)
VoIP gateway?  I'm looking for a good (and hopefully not too expensive) 
VoIP/T.38 gateway for my office. 
Asterisk intergration is not a major factor at this time but may be 
later on.  How well does it work?  Is Echo
a problem? Do the T.38 capablities actually work?  Please share what 
experience you have had.  Also any

experience (good or bad) with other T.38 gateways/ATAs.


Thanks a lot.

Mark C
http://www.psh-inc.com

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[asterisk-users] Hello Everybody, my problem with voicemail.conf

[Ashish Barot]

Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of
voicemail.conf


Actuallt right now i am using Asterisk 1.2 on my LAN environment.
i am able to call all my extension very nicely.

Right now i am trying to deploying voicemail facility for all
extensions, so if anybody is not present, then he/she can leave
message, and that message will be e-mail to particular e-mail id which
i am using in
extension.conf

Upto this moment the voicemail is generating, but it is not e-mail to
any email id. But it comes on
[EMAIL PROTECTED]
that i had check with K-mail application.
Whole message is coming in .wav file extension. on [EMAIL PROTECTED]
also i get few text message from [EMAIL PROTECTED]
on my gmail's spam folder. but in gmail a.c no attachment is coming.

so pl. any body can help me for it

below i am sending my sip.conf , extensions.conf and voicemail.conf

Pl,do the needful.

Thanks.
With Warm Regards,
Ashish Barot.

-
*sip.conf*

[general]
bindport=5060
context=worldbiz
[EMAIL PROTECTED]
nat=yes
allow=all

[]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

[1112]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

[1113]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

-
*extensions.conf

*[general]
static=yes
writeprotect=no


[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])
exten = _111X,3,SendText(Hello I am Ashish Barot here)
exten = _111X,4,Hangup()
exten = _111X,103,SendText(This is my test voice mail message. Try to reply
me)
exten = _111X,104,Hangup()*



voicemail.conf
**
*[general]
attach=yes
[EMAIL PROTECTED]
format=wav
minmessage=0
maxmessage=0


[worldbiz]
 = 1234,Barot,[EMAIL PROTECTED]
1112 = 1234,Ashish Barot,[EMAIL PROTECTED],,attach=yes|format=wav

-

i had search lots of config. files on net from all of them i had prepare
above files. but still not getting it resolve.
so pl. try to reply.

Thanks.
Ashish Barot.
*
**



**
*
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[asterisk-users] asterisk.conf

[Tomislav Parčina]

Why there is no asterisk.conf.sample file?



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] Sangoma card dying after 1hour

[Steve Davies]

Which asterisk versions etc etc?

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:

I am running the newest version, from the sangoma wiki.

Jon

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma card dying after 1hour

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:

 Hello List

 I am having a rather big problem with a sangoma A104 card, I just installed
 to replace a Digium TE410 card, that was acting up.

 But now we have a problem with the sangoma card. It runs great after being
 started, and calls proceed as normal, but after about 1 hour, it stops being
 able to make and receive calls.

 If I run wanpipemon debug,  can see that the card still receives packets
 from the ISDN, but when I make a call, I cant see it in wanpipemon, and
 asterisk just responds with a:

 NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 -
 Circuit/channel congestion)

 I am pretty shure that this is a configuration issue, but are there anything
 I need to be aware of when moving from a Digium card to a sangoma card?

Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded
as there are some resource leak fixes in that version.

Regards,
Steve
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[asterisk-users] Re: asterisk.conf

[Pavel Jezek]

it is in doc/ directory
asterisk-conf.txt


Tomislav Parčina wrote:

Why there is no asterisk.conf.sample file?



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
  



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Re: [asterisk-users] Do I need a CH1 licence for Cisco Phones ?

[Pavel Jezek]

imho, ci$co doesn't support anything other than callmanager as signaling 
server  :-(



Peter Mitchell wrote:

79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.

Whats the non callmanager - SIP licence number for 79X1 ?  

  
  

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RE: [asterisk-users] Sangoma card dying after 1hour

[Asterisk]

Could it be related to this? http://bugs.digium.com/view.php?id=8507

Did you try to contact Sangoma Support? Their replies are prompt.

Regards, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Friday, January 26, 2007 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma card dying after 1hour

Which asterisk versions etc etc?

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
 I am running the newest version, from the sangoma wiki.

 Jon

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
 Sent: 26. januar 2007 10:56
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Sangoma card dying after 1hour

 On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
 
  Hello List
 
  I am having a rather big problem with a sangoma A104 card, I just installed
  to replace a Digium TE410 card, that was acting up.
 
  But now we have a problem with the sangoma card. It runs great after being
  started, and calls proceed as normal, but after about 1 hour, it stops being
  able to make and receive calls.
 
  If I run wanpipemon debug,  can see that the card still receives packets
  from the ISDN, but when I make a call, I cant see it in wanpipemon, and
  asterisk just responds with a:
 
  NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 -
  Circuit/channel congestion)
 
  I am pretty shure that this is a configuration issue, but are there anything
  I need to be aware of when moving from a Digium card to a sangoma card?

 Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded
 as there are some resource leak fixes in that version.

 Regards,
 Steve
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Re: [asterisk-users] Hello Everybody, my problem with voicemail.conf

[Stefan Schmidt]

hello,

maybe you should try adding this to your voicemail configuration.

mailcmd=/usr/sbin/sendmail -t

or whereever your sendmail is located.

then your mails should be send to the wanted adress.

Best regards.

Stefan


Ashish Barot schrieb:

Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of 
voicemail.conf



snipped



Für weitere Fragen stehen wir natürlich gerne unter [EMAIL PROTECTED] oder
059944 - 2010 zur Verfügung.

Mit freundlichen Grüssen
--
Stefan Schmidt
Support/VOIP // [EMAIL PROTECTED] // Tel 059944-2010  //
-
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at  //
-

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[asterisk-users] Asterisk Recording Volume

[george . attopany]

Hi,

I have Asterisk 1.2  + Adit600 Channel bank(which gives analog output and
also takes PSTN lines into the Asterisk system).

Conversations recorded by the ASTERISK  comes in two separate Files:
xx.0-in (GSM Audio) for the  Asterisk Extension Side of the
conversation;

xxx.0-out (GSM Audio) for the Caller's side of the conversation.

I have Quick Time Player to playback the conversations recorded.

Issues I have are:

I am not able to synchronize the  xx.0-in  and  xx.0-out  Files to
playback concurrently.  Each file played gives one side of the
conversation.
I would be grateful for a way out about this.

Secondly, one can hardly hear the  xx.0-out conversation when play
(that
is the incoming conversation can hardly be heard);  whereas the 
xx.0-in
conversation is clearly heard(that is the conversation from the ASTERISK
extension is very loud).

I would appreciate a help out on this as well.

Thanks

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[asterisk-users] strange msg

[Rizwan Hisham]

Hi all,
I dont have any problem, my asterisk is working fine. but on the cli,
asterisk keeps saying  Got SIP response 603 Declined (no dialog) back
from 192.168.0.100. trixbox running on another machine is registered to our
server from address 192.168.0.100. whats the reason of this msg?

--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] TE110P and HDLC problems

[Marc Patino Gómez]

Thanks for your answers :)

I use another server to test  digium board and my config, and it works 
well, so... I think the problem is between chipset, Intel 5000P and 
digium card. I will try to put the digium board in other PCI-X slots, 
and change some timing PCI parameters  in the BIOS.


Regards


Marc Patino Gómez wrote:

Hi!,

this issue makes me crazy. I read a lot of docs, also * mailling list 
and I try a lot of things  without success.


Any help will be appreciated. Here is the info:

Hardware:

Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 5050
Digium TE110P

Software
-
Asterisk version 1.2.12.1
Zaptel version 1.2.8

/etc/zaptel.conf

loadzone=es
defaultzone=es
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

The dammed errors:

Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1

...

I tried the following without success:

- Disable Hyper Threading.
- Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, 
so TE110P has his own IRQ as shows lspci -vb.

- Also I tried with APIC and without APIC.
..


These HDLC errors appear when I physically loop the E1 interface in 
the Card and also appear, and more frequently, when I connect the E1 
circuit (from the Telco) to the interface of the Card.



Thanks a lot




--


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel. Información General: 902 884 633 
Tel. Soporte Técnico: 902 884 622

Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal - 
Netherlands - USA



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Re: [asterisk-users] NAT solutions

[Tim Panton]

On 25 Jan 2007, at 06:57, Brad Templeton wrote:


On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:

In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can manage your hairpin
calls internally, save you bandwidth by trunking the IAX traffic
to the central asterisk and avoid all the NAT hassle by using
a single port (outgoing) and refreshing it often enough for the
router to hold it open.


Tim Panton

www.mexuar.net
www.westhawk.co.uk/


IAX is a fine protocol as far as it goes, however this answer
is really not a workable one.   There are only a few IAX phones,
and they are not nearly as solid and full featured as the many
SIP phones.   There are some IAX termination and origination
providers, but there are far more SIP providers.


I've never had a problem finding an IAX provider indeed
they seem to be more clue'd :-) than the SIP only ones.



For a remote phone, not on the same network as the Asterisk
box (in which event the NAT worries are different) you definitely
want to use the same protocol for the phone as for your
term/orig provider.   Otherwise you will be forced to hairpin
your audio through your asterisk server, adding latency and
wasting bandwidth and cpu for little reason.


Unless you are monitoring calls, want full CDR  etc,
then that's what you want anyway.



In addition, many people just want to do things like give
family or employees a phone they can take home, or take to
a remote location and use on the PBX.   They probably can't
just put up an Asterisk server to make this happen, and
nor should they want to.


I agree. Single SIP phones can usually be got to work behind
a reasonable NAT router.



An additional server is not only more work and requires an
always-on server computer, it's another thing that can go
wrong.


For a single phone - you are quite right. For multiple phones,
I'm not sure I agree - multiple SIP phones behind a NAT router
is going to require some extensive config , or a SIP proxy in the  
router.

If you are going to be maintaining a proxy, why not use asterisk
on an NSlu2 or an WRT ?




No thanks.  Even if you can run Asterisk on a WRT54G, and
thus don't have the $200/year power expense of a server,
it's still not what you really want.

IAX is great but SIP is also a reality, and putting
Asterisk into the just works category is a really
important milestone.  One I think that is intended
to be improved a lot for 1.6.


Ah, but it isn't just asterisk you have to change - it is
all the SIP implementations and all the routers :-)

It will happen, SIP will move such that it uses fewer
ports in a more predictable way (thus becoming more IAX ish)
routers will come with sane SIP proxies etc, but (as I said)
in the meanwhile IAX is a useful tool to have to solve some
of these problems now.



Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] IMAP Voicemail Storage

[David Gomillion]

On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:


On Thursday 25 January 2007 6:30 pm, David Gomillion wrote:
 I mean that I would like to have a system in place so that Asterisk, as
a
 privileged service, can gain access to Courier's IMAP storage. Having to
 keep track of all of our users' passwords in the Asterisk configuration
is
 going to provide a ridiculous amount of administration, as we force them
to
 change their passwords often in our single-sign on environment.

How do they log on to check their voicemail?  Is your SSO system entirely
numeric?

-A.



I'm not talking about setting the voicemail password. I'm talking about not
having to put my users' email passwords in the voicemail.conf file.
Asterisk, if I understand correctly, needs each user's email password to
deliver the voicemail, to integrate messaging into the IMAP server. Or it
needs a general user that has rights to deliver and read any mailbox, which
I don't know of existing in Courier.

You said you had done some testing. What model did you use? Did you put each
user's email username and password in the voicemail.conf, or were you able
to come up with a general user for Asterisk to use when delivering every
voicemail?



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[asterisk-users] Asterisk Recording Volume

[John covici]

I have had no trouble but I record in .wav format and it automatically
mixes it together if I use the *1 or the mixmon app.


on Friday 01/26/2007 [EMAIL PROTECTED]([EMAIL PROTECTED]) wrote
  Hi,
  
  I have Asterisk 1.2  + Adit600 Channel bank(which gives analog output and
  also takes PSTN lines into the Asterisk system).
  
  Conversations recorded by the ASTERISK  comes in two separate Files:
  xx.0-in (GSM Audio) for the  Asterisk Extension Side of the
  conversation;
  
  xxx.0-out (GSM Audio) for the Caller's side of the conversation.
  
  I have Quick Time Player to playback the conversations recorded.
  
  Issues I have are:
  
  I am not able to synchronize the  xx.0-in  and  xx.0-out  Files to
  playback concurrently.  Each file played gives one side of the
  conversation.
  I would be grateful for a way out about this.
  
  Secondly, one can hardly hear the  xx.0-out conversation when play
  (that
  is the incoming conversation can hardly be heard);  whereas the 
  xx.0-in
  conversation is clearly heard(that is the conversation from the ASTERISK
  extension is very loud).
  
  I would appreciate a help out on this as well.
  
  Thanks
  
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]
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Re: [asterisk-users] IMAP Voicemail Storage

[Andrew Kohlsmith]

On Friday 26 January 2007 7:42 am, David Gomillion wrote:
  I'm not talking about setting the voicemail password. I'm talking about
 not having to put my users' email passwords in the voicemail.conf file.
 Asterisk, if I understand correctly, needs each user's email password to
 deliver the voicemail, to integrate messaging into the IMAP server. Or it
 needs a general user that has rights to deliver and read any mailbox, which
 I don't know of existing in Courier.

Ahh; yes that is what I use is the server auth, not user auth.  I will have 
the development machine up this morning; I'll get the config off of it and 
post it here.

 You said you had done some testing. What model did you use? Did you put
 each user's email username and password in the voicemail.conf, or were you
 able to come up with a general user for Asterisk to use when delivering
 every voicemail?

Server authentication under [general]; no individual user passwords.

-A.
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RE: [asterisk-users] NAT solutions

[Ken Williams]

Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one of
the easiest configs to put together.  Works extremely well and requires
opening a single port on each NAT.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Thursday, January 25, 2007 11:19 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] NAT solutions

From: Brad Templeton [EMAIL PROTECTED]
  I have a really dumb question.  It appears that Yahoo, MSN, AIM, you
name
  them, they don't have a NAT problem, and some use SIP.  I don't 
  think
they
  all stay in voice path, either.  What takes?

When you control both ends of the path, you can eliminate all NAT 
problems.  Skype also deals almost perfectly with NAT (by using
other nodes as relays if necessary) as does IAX.   SIP was designed

Thanks for this information.  Does this mean two IAX boxes can talk
behind their respective NAT's (without any server sitting in voice
path)?  I'm imagining this:

Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2

If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.

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[asterisk-users] Dialplan - play sample, interrupt on * and return value?

[Tony Howat]

Hi Asteriskers,

I have the following :

exten = 1,1,Playback(sample)
exten = 1,2,Read(response,,1)
exten = 1,3,GotoIf($[${response} != *]?300:100)
exten = 1,100,Playback(hello)
exten = 1,101, [[[ do stuff ]]]
exten = 1,300,Playback(reject)
exten = 1,301,Hangup

Which plays a confirmation sample, waits for the user to press * and then 
continues with the application or hangs up depending on whether or not * has 
been pressed.


This is fine, but I need to be able to cope with and detect when the user 
presses * during the sample play, and get a return value based on this.


I can't really see a means to do this? Background is concerned with firing 
off to new extensions - it seems doubtful to me that I can have a * 
extension? Read() itself only provides for playing a sample before the read 
with no interruptions.


Am I missing some other way? I'm more familiar with AGI and can see how to 
achieve it using that interface, but I'd rather avoid it for what would be a 
very small bit of dialplan code.


Thanks in advance,

--
Tony

_
Find Love This New Year With match.com! http://msnuk.match.com

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[asterisk-users] Ringing oddity/stupidity

[J. Oquendo]

Anyone experience ring oddities with extensions.conf rollovers? Let me 
summarize...


One of my extensions.conf file is built to ring during the day, ring/go 
to voicemail after a certain time:


[main-aa]
exten = s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1)
exten = s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1)
exten = s,3,Dial(SIP/201,25,tr)
exten = 
s,4,DIal(SIP/211SIP/202SIP/203SIP/209SIP/211SIP/212SIP/213SIP/214,15,tr)

exten = s,5,Background(/etc/asterisk/day)
exten = s,6,Wait(3)
exten = s,7,Voicemail([EMAIL PROTECTED])
exten = s,8,Hangup

[main-night-aa]
exten = s,1,Answer
exten = s,2,Background(/etc/asterisk/night)
exten = s,3,Voicemail([EMAIL PROTECTED])
exten = s,4,Hangup
exten = 1,1,Directory(fakepbxname,internal,l)
exten = 00,1,VoicemailMain([EMAIL PROTECTED])

When in night mode, if someone called, while Asterisk would show the 
phone as ringing (and INDEED the phone would ring) the caller wouldn't 
hear the phone ring. No music, no ringing no thing until the amount of 
time the rings ran out and then be transferred into voicemail. So... 
(un)Leet ASCII explanation:


Caller (after hours) -- Dials in -- Press extension -- Asterisk makes 
transfer -- Caller hears dead air -- No one answers -- Voicemail -- 
Caller now hears voicemail prompts


Asterisk 1.2.13 built by root @ fakepbxname on a i686 running Linux (FC5)

Any thoughts?

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] wireless sip phone with auto answer - are there any

[Jerry Geis]

Does anyone know of a wireless 802.11 sip phone with an auto answer mode?

THanks,

Jerry
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[asterisk-users] Problem solved

[Tony Howat]

Background *does* do what I need, so problem solved.

Thanks to fenlander on #asterisk-uk for the help :)

--
Tony

_
MSN Hotmail is evolving – check out the new Windows Live Mail 
http://ideas.live.com


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RE: [asterisk-users] Sangoma card dying after 1hour

[Jon Schøpzinsky]

Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with 
zaptel 1.2.12 :)

Jon

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 12:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma card dying after 1hour

Which asterisk versions etc etc?

On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
 I am running the newest version, from the sangoma wiki.

 Jon

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
 Sent: 26. januar 2007 10:56
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Sangoma card dying after 1hour

 On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
 
  Hello List
 
  I am having a rather big problem with a sangoma A104 card, I just installed
  to replace a Digium TE410 card, that was acting up.
 
  But now we have a problem with the sangoma card. It runs great after being
  started, and calls proceed as normal, but after about 1 hour, it stops being
  able to make and receive calls.
 
  If I run wanpipemon debug,  can see that the card still receives packets
  from the ISDN, but when I make a call, I cant see it in wanpipemon, and
  asterisk just responds with a:
 
  NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 -
  Circuit/channel congestion)
 
  I am pretty shure that this is a configuration issue, but are there anything
  I need to be aware of when moving from a Digium card to a sangoma card?

 Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded
 as there are some resource leak fixes in that version.

 Regards,
 Steve
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[asterisk-users] Recompiled app_xyz.so and Asterisk Dynamic Loader

[ast guy]

Hi,
I would like to know what is Asterisk Dynamic Loader. Let me
explain what I'm about to ask.

I have three Asterisk servers running my in-house built app_xyz.so
application. Now what I do to save time is compile application on one
server and scp app_xyz.so on rest of servers. All servers have same
OS, H/W specs. Today I checked the logs and observed that at the time
when app_xyz.so was copied on Server-1,2, asterisk restarted. The last
line before restart was Asterisk dynamic loader sort of debug message.
and show uptime was give exact restart time since reloaded.

Now my concern is
1. What about the running sessions at that time,  they were hunged up
by asterisk forcefully ? Because I didn't see the logs of that session
afterwards.
2. Do I really need to restart asterisk gracefully to reload new
modified app_xyz.so application to make changes effective on
production?

Thanks!
-ag
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[asterisk-users] Rxfax and Txfax on Asterisk 1.4

[Remzi Semsettin Turer]

Has anyone successfully installed spandsp and rxfax and txfax applications on 
1.4.0 release of Asterisk?

I tried the latest snapshot of spandsp, as well as couple other previous 
versions. I compiled it fine, downloaded the asterisk.patch, manually patched 
the asterisk files, run .configure, make clean, make menuselect and it shows 
app_txfax and app_rxfax as XX (unavailable).

Each time I made sure no other spandsp versions are installed and put the 
proper path in /etc/ld.so.conf and run ldconfig, prior to compiling Asterisk. 
Still no luck.

Any suggestions?

TIA,

Remzi Turer
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Re: [asterisk-users] Hello Everybody, my problem with voicemail.conf

[Andy Davidson]

On 26 Jan 2007, at 10:43, Ashish Barot wrote:

Upto this moment the voicemail is generating, but it is not e-mail  
to any email id. But it comes on [EMAIL PROTECTED]

[...]

[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])

[...]

[worldbiz]
 = 1234,Barot,[EMAIL PROTECTED]
1112 = 1234,Ashish Barot,[EMAIL PROTECTED],,attach=yes|format=wav



All of your extensions are configured to pass voicemail into mailbox  
 - this is configured to send VM notifications to [EMAIL PROTECTED]  
(i.e. no voicemail is ever hitting mailbox 1112).


-a

--
Regards, Andy Davidson
http://www.devonshire.it/  -  0844 704 704 7  - Sheffield, UK


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Re: [asterisk-users] setting up AMD

[Asterisk]

AT the risk of being rude with a follow up of the same information and a
top post, change the AMDSTATUS of AMD_PERSON to HUMAN.  The example does
not work, if you look at the source for AMD you will see that the status
returned is:

This application sets the following channel variable upon completion:
\n
AMDSTATUS - This is the status of the answering machine
detection.\n
Possible values are:\n
MACHINE | HUMAN | NOTSURE | HANGUP\n
AMDCAUSE - Indicates the cause that led to the conclusion.\n
   Possible values are:\n
   TOOLONG-%d total_time\n
   INITIALSILENCE-%d silenceDuration-%d
initialSilence\n
   HUMAN-%d silenceDuration-%d afterGreetingSilence\n
   MAXWORDS-%d wordsCount-%d maximumNumberOfWords\n
   LONGGREETING-%d voiceDuration-%d greeting\n;

Try changing the tested value.

dave



On Thu, 2007-01-25 at 23:25 -0500, Peter Halliday wrote:
 I already put this in there, but this is the context for the call.  I
 got it right out of voip-info.org's article.  This is correct right?
 
 [outboundmsg1]
 exten = s,1,NoCDR 
 exten = s,n,AMD
 exten = s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach)
 exten = s,n(mach),WaitForSilence(2500)
 exten = s,n,Playback(outboundmsgs/msg1)
 exten = s,n,Hangup
 exten = s,n(humn),WaitForSilence(500) 
 exten = s,n,Playback(outboundmsgs/msg1)
 exten = s,n,Hangup
 
 I'm using broadvoice for the service not sure that it matters.
 
 
 
 On 1/25/07, Matt Florell [EMAIL PROTECTED] wrote:
 I tested it over a SIP channel and an IAX channel and it did
 work, but
 I have not used it in production that way. I only use Zap
 channels(T1
 PRI) In prodution at the locations that I use AMD at.
 
 MATT---
 
 On 1/25/07, Peter Halliday [EMAIL PROTECTED] wrote:
  That's the same code as I have.  It's identical.  Are you
 using it over a
  SIP channel?
  
  Peter
 
 
  On 1/25/07, Matt Florell  [EMAIL PROTECTED] wrote:
   From the VICIDIAL SCRATCH_INSTALL doc:
  
   - cd asterisk-1.2.14/apps
   - wget http://www.eflo.net/files/app_amd2.c
   - mv app_amd2.c app_amd.c
   - vi Makefile
 replace this line(line 32): 
  app_mixmonitor.so app_stack.so
 with this line:
  app_mixmonitor.so app_stack.so app_amd.so
   - wget http://www.eflo.net/files/amd2.conf
   - mv amd2.conf /etc/asterisk/amd.conf
  
   It works with Asterisk 1.2.14 just fine.
  
  
   MATT---
  
  
   On 1/25/07, Peter Halliday  [EMAIL PROTECTED] wrote:
Where can I get the latest copy of this file.  I thought
 google found
ithere, but it doesn't compile correctly on 1.2.14.  And
 the copy on
voip-info.org that I found initially appears to be
 old.  It's not in the
  1.2
tree.
   

   
   
On 1/25/07, Asterisk [EMAIL PROTECTED] wrote:
 On Wed, 2007-01-24 at 12:20 -0800, Michael Collins
 wrote: 
 
 
 
 
 
 
   
 
 
 __ 
  From:[EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On
Behalf Of Peter
  Halliday
  Sent: Wednesday, January 24, 2007 11:56 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] setting up AMD
 
 
  
 
  I'm trying get this working.  I've looked through
 the list, and
  can't
  see how to get AMD to print out more.  I have it
 call and say Hello 
  like I normally would.  I've tried to say more and
 less doesn't seem
  to matter.  After I hangup it does recognize
 hangup.  Here's logging
  during an attempt where I make outbound call and
 answer, but then 
  hangup after 1-2 seconds:
 
  Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
  SIP/sip.broadvoice.com-098c4aa8 6079362172 (null)
 (Fmt: 4) 
  Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD:
 initialSilence
  [8000] greeting [1500] afterGreetingSilence [300]
 totalAnalysisTime
  [5000] minimumWordLength [120] betweenWordsSilence
 [50] 
  maximumNumberOfWords [5] silenceThreshold [256]
  Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto
 destroying call
 
  '[EMAIL PROTECTED] '
  Jan 24 17:01:45 

Re: [asterisk-users] NAT solutions

[Gordon Henderson]

On Thu, 25 Jan 2007, Yuan LIU wrote:

Thanks for this information.  Does this mean two IAX boxes can talk behind 
their respective NAT's (without any server sitting in voice path)?  I'm 
imagining this:


Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2


Using IAX, yes. It's quite straightforward to do. You do need to open the 
IAX port on each NAT device though - this may be called port-forwarding, 
depending on the hardware or its configuration interface. Essentially, you 
port-forward port 4569 from the outside to the IP address of the asterisk 
box on the inside on both sides.


Then have a look at:

http://astrecipes.net/index.php?n=204

To get you going.

Is this the concept of STUN?  Does this also create latency (by adding an 
additional leg in the route), packet loss, even jitter?


STUN doesn't intercept the data. It gives the client device hints as to 
how best to traverse the local NAT firewall.


IAX uses a single port for both commands and data. SIP uses more than one 
and thats when it gets hard as it's easy for a NAT router to track a 
single data stream, but tracking multiple is hard. I have noticed newer 
routers offering SIP NAT traversal though (and the later linux kernels 
claim to be able to do it) I guess, like handling FTP (which also uses 
multiple ports) they are inspecting the SIP packet contents to try to work 
out the RTP ports it's going to use and do the right thing.


I did have issues with a Juniper router recently though - the owner 
claimed it has SIP traversal but it didn't work, but when we turned it off 
and used old fashioned port forwarding it just worked ...


Gordon
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Re: [asterisk-users] Asterisk 1.4 Polycom buddy status

[James Fromm]

Olle E Johansson wrote:


24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:


James Fromm wrote:

The behavior we see is that the SIP interface in the queue will 
sometimes not release from the in-use state.  Connecting to the 
interface from another SIP device and immediately hanging up will 
clear the state.
The phones in question are configured with one line that will except 
only one call.  The device itself does not think it is in-use because 
it will accept another call.  Something in the SIP channel driver is 
not clearing the state when a call is completed.
There is definitely no correlation between this and Asterisk 
restarting.  In fact, if a device is 'stuck' on in-use, restarting 
Asterisk will clear the state.
I've been working on this for a week now.  It only started for us 
because I just implemented the call-limit option in the sip.conf in 
Asterisk for the devices.  See my posts with subject 'Queue and 
Interface time out'.


I believe there is/was a bug relating to call-limit.  Buddy Watch 
doesn't work if you use call-limit and if a call from a queue is 
transfered, the call-limit is not released until the original call is 
terminated.  I do not know if these issues have been fixed or not.


Again, a relation to call transfer. I think the bug is that we don't 
handle call-limits properly during a call transfer. That needs

to be verified and fixed.



There may be, but transfers are not the cause of the issue I describe. 
SIP interfaces that are members of a Queue, will erratically not be 
released from 'in-use' when a call is completed.  I have tested with 
both caller terminated and agent terminated calls and both will cause 
this behavior.  It happens on approximately 20% of all calls the queue 
members receive.  Dialing the SIP device with another device will 
immediately free the status.


I wonder if this only happens on calls sent to the SIP device by the 
Queue application.  I will test that today.

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Re: [asterisk-users] TE110P and HDLC problems

[Marc Patino Gómez]

Problem solved :) :)

I change the OS, I install a Debian Etch x86_32bits and it works 
perfectly with the following software versions:


asterisk 1.2.13
zaptel 1.2.12

so... I don't understand where was the problem. TE110P driver 
version


The previus OS was a Ubuntu 6.10 (codename Edgy) 64bits version.



Marc Patino Gómez wrote:

Thanks for your answers :)

I use another server to test  digium board and my config, and it works 
well, so... I think the problem is between chipset, Intel 5000P and 
digium card. I will try to put the digium board in other PCI-X slots, 
and change some timing PCI parameters  in the BIOS.


Regards


Marc Patino Gómez wrote:

Hi!,

this issue makes me crazy. I read a lot of docs, also * mailling list 
and I try a lot of things  without success.


Any help will be appreciated. Here is the info:

Hardware:

Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 5050
Digium TE110P

Software
-
Asterisk version 1.2.12.1
Zaptel version 1.2.8

/etc/zaptel.conf

loadzone=es
defaultzone=es
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

The dammed errors:

Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1
Jan 25 14:11:52 NOTICE[4934]: chan_zap.c:8316 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 1

...

I tried the following without success:

- Disable Hyper Threading.
- Disable NIC's, RAID controller, usb, serial... to avoid shared IRQ, 
so TE110P has his own IRQ as shows lspci -vb.

- Also I tried with APIC and without APIC.
..


These HDLC errors appear when I physically loop the E1 interface in 
the Card and also appear, and more frequently, when I connect the E1 
circuit (from the Telco) to the interface of the Card.



Thanks a lot







--


Marc Patino Gómez
Dpto. Sistemas

Claranet España. Servicios Internet
C/General Almirante 2-28, Torres Cerdá
08014 Barcelona
Tel. Información General: 902 884 633 
Tel. Soporte Técnico: 902 884 622

Fax: +34 93 445 19 20
www.claranet.es

Claranet Group: United Kingdom - Spain - France - Germany - Portugal - 
Netherlands - USA



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Re: [asterisk-users] Adding 4 more POTS lines

[Drew Gibson]

cb wrote:

On Jan 25, 2007, at 5:38 PM, Leif Neland wrote:

A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a 
TDM404B fully populated 4FXO card.


I'm currently testing a GXW-4108... my verdict is still out. I've had 
some problems, some minor, some major.


In the minor department, it does not always reboot when instructed to 
via the web interface. I think I've tracked it to the reboot button on 
a regular screen is ignored, but the reboot from the post update 
screen goes thru. This is likely a minor bug in the firmware.



FWIW I've seen this in the HandyTone 386 too


Also in the not so minor category, there doesn't appear to be any easy 
way of backing up the config files. When it polls the tftp server on 
boot, it does look for a config file, but since there doesn't appear 
to be any way to save one out of the unit, and no documentation or 
otherwise (that I've found) to create one from scratch... it makes it 
very difficult to save settings and then easily restore them.



Try http://www.grandstream.com/y-configurationtool.htm
You can get the option numbers and values from the source html of the 
web page. (I am assuming the GXW-4108 works the same as other 
Grandstream products)



--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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Re: [asterisk-users] NAT solutions

[Julio Arruda]

Gordon Henderson wrote:

On Thu, 25 Jan 2007, Yuan LIU wrote:

Thanks for this information.  Does this mean two IAX boxes can talk 
behind their respective NAT's (without any server sitting in voice 
path)?  I'm imagining this:


Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2


Using IAX, yes. It's quite straightforward to do. You do need to open 
the IAX port on each NAT device though - this may be called 
port-forwarding, depending on the hardware or its configuration 
interface. Essentially, you port-forward port 4569 from the outside to 
the IP address of the asterisk box on the inside on both sides.


Then have a look at:

http://astrecipes.net/index.php?n=204

To get you going.

Is this the concept of STUN?  Does this also create latency (by adding 
an additional leg in the route), packet loss, even jitter?


STUN doesn't intercept the data. It gives the client device hints as to 
how best to traverse the local NAT firewall.


IAX uses a single port for both commands and data. SIP uses more than 
one and thats when it gets hard as it's easy for a NAT router to track a 
single data stream, but tracking multiple is hard. I have noticed newer 
routers offering SIP NAT traversal though (and the later linux kernels 
claim to be able to do it) I guess, like handling FTP (which also uses 
multiple ports) they are inspecting the SIP packet contents to try to 
work out the RTP ports it's going to use and do the right thing.


I did have issues with a Juniper router recently though - the owner 
claimed it has SIP traversal but it didn't work, but when we turned it 
off and used old fashioned port forwarding it just worked ...


My experience with SIP ALG implemented in several routers/modems/NAT 
box/fillintheblanksis not exactly good :-)
I saw many cases where the messing around done by the middlebox break 
either authentication+integrity or even the voice path.
I've not tried the SIP ALG in the iptables modules, but, not sure how 
much better would be :-)..




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Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian

[Facundo Ameal]

Moises,
  I 've stated testing by raising all timers a bit. Everything went
worse, now there are more failed calls. Can you give me a hint of
which timers to modify and, if you know, a more extensive explanation
of each one? I know it's documented into the file but I cannot catch
the concept.

Thanks you very much!

Greets.

On 1/21/07, Facundo Ameal [EMAIL PROTECTED] wrote:

Thanks Moises, I was trying to find some consistence, but the only
similarity I could find is that much of the calls that fail are long
distance ones or international. It fails in both, Telmex and Meridian
link.
I 'll try looping.

I'll be posting results soon. I hope I can manage to get it work.

Thanks for your help.

Regards.

On 1/19/07, Moises Silva [EMAIL PROTECTED] wrote:
 Similar probles I had were fixed incrementing one of the timers, but
 if you have already done that, I have no idea of your problem, you
 require to debug the problem and try to find some consistence in the
 failures, find if the failure is on the Asterisk - telco Link, or in
 the Asterisk - meridian link? find if putting in loop your own
 asterisk still fails, etc etc.

 Kind Regards

 On 1/18/07, Facundo Ameal [EMAIL PROTECTED] wrote:
  Thanks for your help, but I've already adjusted timers on the source
  code. I found your document a week ago and read it.
  Do you really think that is a matter of timers only?
 
  Greets!
 
  On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote:
   Sometimes timers need to be adjusted on the mfcr2 source code.
   Sometimes is missconfiguration. Anyway, may be this document can help
   you out to debug the problem:
  
   http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
  
   Kind Regards
  
   On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote:
Hi everyone!
I'm having some issue trying to place calls with asterisk connected to
an E1 R2 from Telmex Argentina. The other E1 port is connected to a
Meridian which also uses R2 protocol. Calls sometimes fail with
different error messages such as: Unicall protocol error 32773, 32772,
32769. Some other calls fail saying:
   Far end disconnected(cause=Destination out
of order [27])
   Far end disconnected(cause=User alerting,
no answer [19])
   Far end disconnected(cause=Switching
equipment congestion [42])
   Far end disconnected(cause=User busy [17])
   
I don't think those causes are real, because if you use another line,
yo establish the call. Could it be something about timing of ABCD
bits?
   
I'm using:
Asterisk 1.2.6
Zaptel 1.2.5
libmfcr2 0.0.3
libunicall 0.0.3
libsupertone 0.0.2
spandsp-0.0.3
   
And this is my unicall.conf:
   
[channels]
loglevel=1023
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callerid=asreceived
callreturn=yes
echocancel=no
echocancelwhenbridged=no
echotraining=no
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
   
musiconhold=default
protocolclass=mfcr2
protocolvariant=ar,10,4,15
protocolend=cpe
group=1
context=from-zaptel
channel = 1-15
channel = 17-29
   
loglevel=0
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callerid=asreceived
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
   
protocolclass=mfcr2
protocolvariant=ar,0,12,12
protocolend=cpe
group=2
context=hacia-afuera
channel = 32-46
channel = 48-60
   
   
Thanks in advance!
   
Greets!
   
   
   
--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
   
Share your knowledge, use free software.
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   --
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http://www.gnu.org;
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  --
  Facundo Ameal.
  famealatgmaildotcom
  Linux User #395088
 
  Share your knowledge, use free software.
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 --
 Su 

[asterisk-users] International Carriers

[Facundo Ameal]

Hello everyone!
I 've looking for carriers which can terminate my international calls.
They must accept payments from Argentina and give me interconection to
my Asterisk. I'd appreciate your help or recomendations.


Regards.

--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

Share your knowledge, use free software.
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Re: [asterisk-users] Zap channels staying offhook - restart required

[Shane Spencer]

Just for giggles can you set an absolute timeout in the dialplan for
all calls in and out of that span?

On 1/25/07, kjcsb [EMAIL PROTECTED] wrote:

I have a situation where the two Zap channels on a TDM400 are staying
offhook after a random period of time; it is not (I believe) related to the
FXO side not hanging up. Actually I suspect the server is overheating but I
need to do more analysis.

Anyway, my question is, how do I get the offhook status to reset? So far
only a server reboot works. I tried:
- physically disconnecting the line from the socket
- restarting asterisk
- zap destroy channel and restarting asterisk

Any suggestions on how to avoid a reboot?

Also suggestions on debugging this would be appreciated.

Regards

Cameron

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Re: [asterisk-users] International Carriers

[Rafael Canchola]

Hi:
I am working in a VoIP Carrier Company, I could provider you the service
for your internationals calls. 
Please visit

www.fonetglobal.com and call me, my phone number is +52 442 167 08 00
x214 Rafael Canchola.
Thanks.
At 09:54 a.m. 26/01/2007, Facundo Ameal wrote:
Hello everyone!
I 've looking for carriers which can terminate my international
calls.
They must accept payments from Argentina and give me interconection
to
my Asterisk. I'd appreciate your help or recomendations.

Regards.
-- 
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
Share your knowledge, use free software.
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RafaelCanchola
Product Development Engineer,

FonetGlobal Inc.
[EMAIL PROTECTED] 

http://www.fonetglobal.com
Ph.
+ 52 800 022 10 21 ext. 214
 + 52 442 167 08 00
VoIP
523663899
d00d!
cyberalph



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Re: [asterisk-users] Semi OT - Point to Point FXO/FXS GatewayCommunication

[C F]

On 1/26/07, Yuan LIU [EMAIL PROTECTED] wrote:

From: C F [EMAIL PROTECTED]

Cory, it's called dialplan magic it realy depends what PBX it is, not
all of them allow dial plan magic. But it is possible on most pbxes.

CF: What exactly is diaplan magic?  I googled but found little info.


Did you really? You made my day.



The basic use case in Cory's posting does not seem to require special
programming in PBX, if my understanding is correct:

phones 1,2,3  --- (FXS' 1,2,3)PBX(FXS' 4,5,6) --- (A-FXO's
4,5,6)Asterisk A
 | { IP } |
   Asterisk B(B-FXS' 4,5,6) --- phones 4,5,6

In this case, Asterisks A and B only need to agree on sending the same
signals received by A-FXO 4 (which always come from PBX-FXS 4) to B-FXS 4
(onto phone 4), and sending the same numbers received by B-FXS 4 (from phone
4) to PBX-FXO4 (via A-FXO 4) and so on.  PBX would have no knowledge that
it's not talking to a POTS phone.  Is this correct?


He is not talking about 2 Asterisks, but about legacy PBX systems. In
most cases the problem is that an FXS gateway using VoIP to an FXO
gateway will not be able to send a flash down VoIP to the FXO, which
is needed for the legacy PBX features. Thats why hooking into on
asterisk connected using FXO localy to the legacy system, and VoIP
extensions on asterisk will work much better than a remote asterisk
system. This is where DP magic might come in, but could work without
any modifications to the legacy PBX dialplan.

BTW, dialplan magic means that you create special dialplan that based
on the digits routes it differently, I think I made up the term (if
it's a term) and google might not yet have picked up on it.



Yuan Liu

On 1/24/07, Cory Andrews [EMAIL PROTECTED] wrote:
Has anyone had any experience using FXO and FXS gateways to extend
legacy PBX extensions to remote users?  I have a customer who needs to
do this, but wants seamless, two way communication, with a SIP server
and without the need for 2-stage dialing.  If anyone has any experience
with a solution please let me know.

Thanks

Cory Andrews


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[asterisk-users] TDM2401 (FXO) Hangup

[C F]

Anybody else having trouble with hangup detection on the TDM2400 FXO
modules? It works most of the time, but sometimes I get hung lines in
Voicemail. And only with the TDM24xx not with the TDM400 or Adit 600
CB. Anybody else have this?
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Re: [asterisk-users] NAT solutions

[Tim Panton]

On 26 Jan 2007, at 06:19, Yuan LIU wrote:


From: Brad Templeton [EMAIL PROTECTED]
 I have a really dumb question.  It appears that Yahoo, MSN, AIM,  
you name
 them, they don't have a NAT problem, and some use SIP.  I don't  
think they

 all stay in voice path, either.  What takes?

When you control both ends of the path, you can eliminate all NAT
problems.  Skype also deals almost perfectly with NAT (by using
other nodes as relays if necessary) as does IAX.   SIP was designed


Thanks for this information.  Does this mean two IAX boxes can talk  
behind their respective NAT's (without any server sitting in voice  
path)?  I'm imagining this:


Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2

If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.


Yes, with 1 proviso - one end needs a known IP address and a port map
for udp 4569 in the router. The other can simply register to it with  
zero

router config.


Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Best way to connect analog modem

[Tim Panton]

On 25 Jan 2007, at 01:40, Bastian Schern wrote:


Hello Asterisk fans,

I try to connect an analog modem to Asterisk. The modems are connected
e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm
using a Wildcard TE110P (E1).

Is it possible to connect the modems to an ATA?
Which ATA I should use for that scenarios?


It is possible, but I really wouldn't use it for an alarm system,
modem over VoIP is unreliable.

I have had it 'working' with a sipura ata  and an E1 PRI over a
quiet lan segment.

I say 'working' because the most I could get out of the modem
with any reliability was 9600 baud - any of the higher speed modulations
failed miserably (and slowly).

If your analog modems are 1200 baud you should be ok :-)

Better get a device to sit in front of the asterisk and have it
split off a couple of analog channels before they reach it.

Tim.


Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Asterisk forgetting about client registration or Polycom phone forgetting to register?

[Marco Mouta]

check register expiration on polycom , probably is higher than 3600 sec
(default on asterisk) , so after this 3600 , imagine polycom as an expire of
6000sec, there's a gap of 2400sec that polycom isn't registred!

On 12/10/06, C F [EMAIL PROTECTED] wrote:


While what you say might/should help, it doesn't fix the problem.

Additional information, since posting this question, till now,
everything worked fine, since it wasn't a working day and only 3
manager sessions are open to asterisk, I'm suspecting that it has to
do either if there are lots of phone calls going on, or when the
manager has more than 3 active connections.

On 12/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
  I'm having trouble with Polycom 501 phones that asterisk forgets how
  to reach them.
 ...
  host=dynamic

 We've found much better results with the static IP here.

 Can you try this?

 --
 Henry J. Cobb
 http://www.io.com/~hcobb/

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Re: [asterisk-users] WellTech 380x Gateway

[Vlad B]

try to use sipura SPA2102. it has T38 and works well with WellGate 5250 adn 
cisco 53xx on other end.

Hope this helps...

Vlad
- Original Message - 
From: Mark Coccimiglio [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 3:27 AM
Subject: [asterisk-users] WellTech 380x Gateway



Ok this is a simple question...

What has been your experience with the WellTech 38xx series (I'm looking 
specifically at the 3802)
VoIP gateway?  I'm looking for a good (and hopefully not too expensive) 
VoIP/T.38 gateway for my office. Asterisk intergration is not a major 
factor at this time but may be later on.  How well does it work?  Is Echo
a problem? Do the T.38 capablities actually work?  Please share what 
experience you have had.  Also any

experience (good or bad) with other T.38 gateways/ATAs.


Thanks a lot.

Mark C
http://www.psh-inc.com

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Re: [asterisk-users] Asterisk 1.4 Polycom buddy status

[Olle E Johansson]

26 jan 2007 kl. 16.31 skrev James Fromm:


Olle E Johansson wrote:

24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:

James Fromm wrote:

The behavior we see is that the SIP interface in the queue will  
sometimes not release from the in-use state.  Connecting to the  
interface from another SIP device and immediately hanging up  
will clear the state.
The phones in question are configured with one line that will  
except only one call.  The device itself does not think it is in- 
use because it will accept another call.  Something in the SIP  
channel driver is not clearing the state when a call is completed.
There is definitely no correlation between this and Asterisk  
restarting.  In fact, if a device is 'stuck' on in-use,  
restarting Asterisk will clear the state.
I've been working on this for a week now.  It only started for  
us because I just implemented the call-limit option in the  
sip.conf in Asterisk for the devices.  See my posts with subject  
'Queue and Interface time out'.


I believe there is/was a bug relating to call-limit.  Buddy Watch  
doesn't work if you use call-limit and if a call from a queue is  
transfered, the call-limit is not released until the original  
call is terminated.  I do not know if these issues have been  
fixed or not.
Again, a relation to call transfer. I think the bug is that we  
don't handle call-limits properly during a call transfer. That needs

to be verified and fixed.


There may be, but transfers are not the cause of the issue I  
describe. SIP interfaces that are members of a Queue, will  
erratically not be released from 'in-use' when a call is  
completed.  I have tested with both caller terminated and agent  
terminated calls and both will cause this behavior.  It happens on  
approximately 20% of all calls the queue members receive.  Dialing  
the SIP device with another device will immediately free the status.


I wonder if this only happens on calls sent to the SIP device by  
the Queue application.  I will test that today.


If you are using chan_agent as a proxy channel, check if that changes  
things.


/O
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Re: [asterisk-users] Adding 4 more POTS lines

[cb]

On Jan 26, 2007, at 10:39 AM, Drew Gibson wrote:

You can get the option numbers and values from the source html of  
the web page. (I am assuming the GXW-4108 works the same as other  
Grandstream products)


I'll try that out, thanks!

I did see a thread on another forum mentioning the HTML source for  
other products would yield the info needed, but without some kind of  
an example, I didn't know how exactly to turn that into what was needed.


I'd assume, since the 4108 web setup looks and acts exactly like my  
Grandstream GXP-2000 phone, that the same trick would work for all  
their products. I'm certainly going to give it a try.


Someone from Grandstream reached out to me last night after seeing my  
review post about the 4108, so when I give him a call in a little  
while, I'll double check that their config tool will in fact work  
with it as well.


-chris
www.mythtech.net


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[asterisk-users] Analog FXO status checking

[François Delawarde]

Hi all,

I would like to make a script/program that would be able to show lots of 
status information from my analog FXO lines (and FXS lines in the near 
future).


Example of interesting status information:
- Hook status: is there a call being made with that zap?
- Voltage status: cable connected, voltage values (if possible), line 
ringing?

- RX/TX Volume status

I'm using a TDM400 card with FXO modules plugged on spanish lines, and 
trying to parse zap show channel X with Asterisk 1.4.0 gives me 
unexpected results:


- Hookstate (FXS only) line shows Onhook when cable is disconnected 
and Offhook when cable is connected, whether there is a phone call or 
not... Is it normal (with an FXO line)? Can someone explain me why?


- The only way i found to know if the line is in use is to check the 
Echo Cancellation line which shows currently ON... Is there another 
way without having to parse core show channels concise?


- No volume information, voltage status or anything.

Would it be possible to check status of those lines (with voltage info, 
...) making a program that would read information from /dev/zap/X 
character devices (ioctl?) without having to stop asterisk (/dev/zap/3: 
Device or resource busy when asterisk is running)?


Thanks in advance for anyone that could help me!
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[asterisk-users] TDM400P with FXS module problem

[Franz Wu]

Hi list

I conncte my Panasonic KX-TCD705TW cordless phone to TDM400P FXS module.
When the system boots or reboots, the LED on the backlit of TDM400P OFTEN
gets off and dmesg shows problem with FXS module, as follows


Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.0 Echo Canceller: MG2
ACPI: PCI interrupt :02:08.0[A] - GSI 19 (level, low) - IRQ 177
Freshmaker version: 71
Freshmaker passed register test
Timeout waiting for calibration of module 0
Timeout waiting for calibration of module 0
Proslic Failed on Second Attempt to Auto Calibrate
Proslic Failed on Second Attempt to Calibrate Manually.
(Try -DNO_CALIBRATION in
Makefile)
Module 0: FAILED FXS (FCC)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)
Registered tone zone 14 (Taiwan)

When this occurs, no dialtone is heard after off-hook.
I have to rmmod wctdm, unplug the phone cable from TDM400P, modprobe wctdm,
and plug phone cable to make this line work. The point is ztcfg without
rmmod wctdm cannot make working.

I run asterisk 1.4.0 / zaptel 1.4.0  on Debian 3.1r4. I compile from source.

Any help or recommendation will be appreciated.

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[asterisk-users] h323 compile error

[Jerry Geis]

I am getting the following compile error on h323.
Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12
pwlib 1.5.2 and openh323 1.12.2

I have pwlib compiled and installed.
I have openh323 compiled and installed.

I went in the channels/h323 directory and did make opt

What shall I do?

Jerry


../../include/asterisk/utils.h: In function `void
ast_slinear_saturated_divide (short int *, short int *)':
../../include/asterisk/utils.h:199: warning: `always_inline' attribute 
directive ignored

../../include/asterisk/utils.h: In function `int inaddrcmp (const
sockaddr_in *, const sockaddr_in *)':
../../include/asterisk/utils.h:217: warning: `always_inline' attribute 
directive ignored

In file included from ast_h323.cxx:51:
ast_h323.h: At top level:
ast_h323.h:159: type specifier omitted for parameter
ast_h323.h:159: parse error before `*'
ast_h323.cxx:957: type specifier omitted for parameter
ast_h323.cxx:957: parse error before `*'
ast_h323.cxx: In method `H323Channel
*MyH323Connection::CreateRealTimeLogicalChannel (...)':
ast_h323.cxx:959: `capability' undeclared (first use this function)
ast_h323.cxx:959: (Each undeclared identifier is reported only once for
each function it appears in.)
ast_h323.cxx:959: `dir' undeclared (first use this function)
ast_h323.cxx:959: `sessionID' undeclared (first use this function)
make: *** [ast_h323.o] Error 1
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Re: [asterisk-users] Thomson ST2030S and BLF

[Alberto Pastore]

Andrew Joakimsen ha scritto:
I know of the call pickup issues but what asterisk issue and what BLF 
issue?


On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:

Andrew Joakimsen ha scritto:
 Actually I noticed just three days ago there is a new release, and the
 releae notes seem to address

 Disable TrMail and Pickup keys
 Disable call progress indication
 ___
but it does not address poor guys' troubles with asterisk, blf and
call pickup...



If you configure a thomson key to supervise a line,
with the proper hints in extensions.conf,
BLF works great.

Unfortunately,
if you press, by mistake or choice, a
flashing key (when the related sip extension is ringing):

1-you won't pickup the call, as it fails
2-the key will remain flashing and useless until you reboot
 the phone



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[asterisk-users] PHP AGI script callerid question

[Michelle Dupuis]

I am trying to set callerid from a PHP script, using one of two functions as
shown below (setid1 and setid2).  The first function works great with
regular names and numbers, BUT, if I call the function with
(Test,UnknownNumber), the cid number gets set to asterisk.  Why is my
passed number parameter not being accepted in this case?
 
The second function uses the new/recommended method of setting cid name and
number, but it has NO EFFECT.  (i.e. the name and number remain at the
asterisk default).  Why is this one not working?
 
Thanks,
MD
 
==
 
 
// Test #1
function setid1($name,$number) {
  $newid =  \  . trim( substr( trim( $name ), 0, 15 ) )
   . \ . trim( substr( str_replace(  , , $number ), 0, 24 )
)
   .;
  obj-set_callerid( $newid );
}
 
// Test #2
function setid1($name,$number) { 
$obj-cmd(Set(CALLERID(name)=\$name\);
$obj-cmd(Set(CALLERID(num)=\$number\); 
}
 
 
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[asterisk-users] Asterisk on IBM NEBS compliant Blade Server

[Ahsan Masood]

 

Hi All,

 

Asterisk on IBM NEBS compliant Blade Server sounds great. 

 

There is some information at
http://www.voip-info.org/wiki/view/Asterisk+hardware#IBMNEBScompliantBla
deServerforTelcoappli

 

I couldn't find further details on this,  Have some one used this ? or
have any details on this ?

 

Regards,

 

Ahsan

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RE: [asterisk-users] NAT solutions

[Yuan LIU]

From:"Ken Williams" [EMAIL PROTECTED]Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one ofthe easiest configs to put together.Works extremely well and requiresopening a single port on each NAT.
Now I realize that I took the wrong assumption that all NAT traversal is blind traversal. By "blind" Ipicture no port forwarding or any special config - no UPNP, either. At leastblind in Linksys grade equipment where few people would add more restrictive rules. This is the area that FWD, MSN, Yahoo shine. (No way to use any of these in my company's network.) When people talk about "just works" category, I think of this scenario.
Any info about Peerio and possibility to have peerio with Asterisk?
Yuan Liu

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Re: [asterisk-users] setting up AMD

[Peter Halliday]

I downloaded version 1.4.0 compiled and installed it.  This is my
extensions.conf:

[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,NoOp(${AMDSTATUS})
exten = s,n,NoOp(${AMDREASON})
exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten = s,n,Playback(outboundmsgs/msg1)
exten = s,n,Hangup

I tried echoing out the values to ensure they were correct.  When I tried
the log I got this results below, which shows that it does not ever return
from AMD function.  I pumped the logging up to 6, with no real difference in
what is being displayed.  If I just wait until the timeout period expired it
doesn't have any impact either.  I actually have to hang up before AMD
returns and when I do hangup it doesn't output anything anyway.


   Channel SIP/sip.broadvoice.com-08f24a68 was answered.
   -- Executing [EMAIL PROTECTED]:1] NoCDR(SIP/sip.broadvoice.com-08f24a68,
) in new stack
[Jan 26 13:32:34] NOTICE[18188]: cdr.c:424 ast_cdr_free: CDR on channel
'SIP/sip.broadvoice.com-08f24a68' not posted
[Jan 26 13:32:34] NOTICE[18188]: cdr.c:426 ast_cdr_free: CDR on channel
'SIP/sip.broadvoice.com-08f24a68' lacks end
   -- Executing [EMAIL PROTECTED]:2] AMD(SIP/sip.broadvoice.com-08f24a68,
) in new stack
   -- AMD: SIP/sip.broadvoice.com-08f24a68 55 (null) (Fmt: 4)
   -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800]
totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50]
maximumNumberOfWords [3] silenceThreshold [256]
   -- AMD: HANGUP
[Jan 26 13:33:08] DEBUG[18188]: pbx.c:2383 __ast_pbx_run: Extension s,
priority 2 returned normally even though call was hung up
[Jan 26 13:33:08] NOTICE[18188]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/[EMAIL PROTECTED]
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[asterisk-users] ATCOM AT 468 manuals and firmware anyone?

[Erick Perez]

Hi there, im looking for another place that provides manuals and
firmware updates for the ATCOM AT 468 and their configuration with
asterisk.
the site www.atcom.com.cn has non functional download links.

I have several of these units but it came only with one CD, I
misplaced it and I cant remember how to factory reset them and what
will be the default password in the GUI.

thanks for your help.


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] Analog FXO status checking

[Tzafrir Cohen]

On Fri, Jan 26, 2007 at 06:17:03PM +0100, François Delawarde wrote:
 Hi all,
 
 I would like to make a script/program that would be able to show lots of 
 status information from my analog FXO lines (and FXS lines in the near 
 future).
 
 Example of interesting status information:
 - Hook status: is there a call being made with that zap?
 - Voltage status: cable connected, voltage values (if possible), line 
 ringing?
 - RX/TX Volume status
 
 I'm using a TDM400 card with FXO modules plugged on spanish lines, and 
 trying to parse zap show channel X with Asterisk 1.4.0 gives me 
 unexpected results:
 
 - Hookstate (FXS only) line shows Onhook when cable is disconnected 
 and Offhook when cable is connected, whether there is a phone call or 
 not... Is it normal (with an FXO line)? Can someone explain me why?
 
 - The only way i found to know if the line is in use is to check the 
 Echo Cancellation line which shows currently ON... Is there another 
 way without having to parse core show channels concise?
 
 - No volume information, voltage status or anything.

What about ztdiag? Is it useful?

It is not robust for usage with scripts, as its output goes to the
kernel. It also requires a code change as that ioctl is disabled by
default for reasons of stack size. If you're interested, I believe it
can be written so it on't keep data on the stack.

(BTW: users of the Xorcom Astribank have much of this information
already available under /proc/xpp )

 
 Would it be possible to check status of those lines (with voltage info, 
 ...) making a program that would read information from /dev/zap/X 
 character devices (ioctl?) without having to stop asterisk (/dev/zap/3: 
 Device or resource busy when asterisk is running)?

No problem, basically. Take a look at, e.g. ztmonitor.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] setting up AMD

[Peter Halliday]

Of note, I tried the same call using IAX2 instead of SIP, and it was fine.
This may either be 1) a configuration problem or 2) a SIP provider problem.
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[asterisk-users] Asterisk dropping audio

[Edoardo Serra]

Hi all,

I have a problem with Asterisk dropping audio.
While in call, audio gets dropped for a while (from 5 to 60 secs, and 
obviously users often hangup, this means that I'm not sure the audio is 
always coming back after 60 secs), in the meantime the call remains up 
and no SIP signalation is generated.


It happens randomly so it's very difficult to debug.
I cannot see common circumstances when it happens (load average is 
always between 0.10 and 0.95, concurrent calls from 1 to 60 on a 2xXeon 
3GHz with 2GB RAM).


Calls are terminated to PSTN via other Asterisks with E1 (IAX2) or via 
SIP to other VoIP carriers.
That problem happens with every different termination randomly, it also 
happens with calls between our users.
(Well... I cannot exclude it's a termination problem, but I cannot find 
a common way to reproduce it)


I'm using Asterisk 1.2.13 with res_perl (used to do lcr and to post 
customized cdr to mysql)

I also tried 1.2.14 without solving that issue
Kernel is a 2.6.18 vanilla on a linux gentoo

I have g729 codec from digium installed and licensed, there are enough 
licenses available (I was tihinking of an issue of codec but I'm not 
sure it happens only with g729 calls)
I now installed free g729 to see if it helps but I don't have any 
feedback yet


I have an OpenSER acting as a load balancer for 2 asterisks but I don't 
think it could be responsible for that (I'm not using any kind of RTP 
proxy, rtp stream goes directly from user to asterisks)


Every kind of help is really appreciated

Regards

Edoardo Serra
WeBRainstorm S.r.l.

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[asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

[Ricardo Carvalho]

Dear all,

How may I configure my extensions.conf so that only the boss's secretary 
can call the boss through his extension, all others when dial his 
extension only makes the boss's secretary phone ring, not his. If she 
wants, she can transfer the incoming call to the boss dialling his 
extension.


I've tried the following, but it doesn't work:

exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
exten = _boss_extension,1,Dial(SIP/secretary_extension)

This doesn't work because when the secretary tries to transfer the call 
to the boss (using her phone's transfer key, not #), one REFER SIP 
message is sent back to the caller's phone providing him the new address 
for whom the next INVITE should be sent. That INVITE is sent, but when 
reaches Asterisk, that INVITE matches this line:


exten = _boss_extension,1,Dial(SIP/secretary_extension)

and not this one:

exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)



Any ideas of how may I solve this issue?
Regards,
Ricardo.
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RE: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

[Jonathan k. Creasy]

Why don't you just give the secretary the boss' REAL extension and give a 
different extension to the world that just rings the secretary? 

-jonathan

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
 Sent: Friday, January 26, 2007 12:13 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Only secretary can call the boss, all others
 only reach the secretary when dial the boss extension
 
 Dear all,
 
 How may I configure my extensions.conf so that only the boss's secretary
 can call the boss through his extension, all others when dial his
 extension only makes the boss's secretary phone ring, not his. If she
 wants, she can transfer the incoming call to the boss dialling his
 extension.
 
 I've tried the following, but it doesn't work:
 
 exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
 exten = _boss_extension,1,Dial(SIP/secretary_extension)
 
 This doesn't work because when the secretary tries to transfer the call
 to the boss (using her phone's transfer key, not #), one REFER SIP
 message is sent back to the caller's phone providing him the new address
 for whom the next INVITE should be sent. That INVITE is sent, but when
 reaches Asterisk, that INVITE matches this line:
 
 exten = _boss_extension,1,Dial(SIP/secretary_extension)
 
 and not this one:
 
 exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
 
 
 
 Any ideas of how may I solve this issue?
 Regards,
 Ricardo.
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 --
 No virus found in this incoming message.
 Checked by AVG Free Edition.
 Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007
 11:11 AM
 

-- 
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Checked by AVG Free Edition.
Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007 
11:11 AM
 
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[asterisk-users] Polycom Provistioning Issue

[Jason Walker]

From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing, 
it errors out  with a 0x1 error


Any Ideas?


1005195711|so   |4|00|-- Initial log entry --
1005195711|so   |4|00|+++ Note that bootrom log times are in GMT +++
1005195711|hw   |4|00|Initial log entry.
1005195711|wdog |4|00|Initial log entry
1005195711|cfg  |4|00|Initial log entry
1005195711|copy |3|00|Initial log entry
1005195711|cdp  |4|00|Initial log entry
1005195711|cdp  |5|00|CDP is DISABLED.
1005195711|cdp  |5|00|802.1Q/VLAN tagging is DISABLED.
1005195711|so   |3|00|Platform: Model=SoundPoint IP 501, 
Assembly=2345-11500-040 Rev=A

1005195711|so   |3|00|Platform: Board=2345-11500-040 A
1005195711|so   |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, 
Subnet Mask=255.255.255.0

1005195711|so   |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08
1005195711|so   |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 
24-Aug-06 18:05

1005195711|so   |3|00|Application, main: P/N=3150-11069-322
1005195711|app1 |4|00|Initial log entry.
1005195711|app1 |3|00|DNS resolver server is '192.168.15.10'
1005195711|app1 |3|00|DNS resolver search domain is ''
1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash 
e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= 
tn=CircaIP

1005195712|so   |3|00|Link status is Net up Speed 100 full Duplex, PC down.
1005195722|cfg  |3|00|Beginning to provision phone
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from 
'192.168.15.52'

1005195722|cfg  |3|00|Image bootrom.ld has not changed
1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 
(addr 1 of 1)
1005195722|cfg  |3|00|Downloaded bootROM is identical to Current version 
3.2.2
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from 
'192.168.15.52'
1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on 
attempt 1 (addr 1 of 1)
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from 
'192.168.15.52'

1005195724|cfg  |3|00|Image sip.ld has not changed
1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 
1 of 1)
1005195724|cfg  |3|00|Downloaded application image is identical to 
current version

1005195724|cfg  |3|00|Phone successfully provisioned
1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
1005195755|app1 |4|00|Loaded application sip.ld successfully, errors 0x0.
1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55 2006


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RE: [asterisk-users] Polycom Provistioning Issue

[Darryl Dunkin]

This is typically an error in one of your config files, either
0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look
like?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue

 From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,

it errors out  with a 0x1 error

Any Ideas?


1005195711|so   |4|00|-- Initial log entry --
1005195711|so   |4|00|+++ Note that bootrom log times are in GMT +++
1005195711|hw   |4|00|Initial log entry.
1005195711|wdog |4|00|Initial log entry
1005195711|cfg  |4|00|Initial log entry
1005195711|copy |3|00|Initial log entry
1005195711|cdp  |4|00|Initial log entry
1005195711|cdp  |5|00|CDP is DISABLED.
1005195711|cdp  |5|00|802.1Q/VLAN tagging is DISABLED.
1005195711|so   |3|00|Platform: Model=SoundPoint IP 501, 
Assembly=2345-11500-040 Rev=A
1005195711|so   |3|00|Platform: Board=2345-11500-040 A
1005195711|so   |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, 
Subnet Mask=255.255.255.0
1005195711|so   |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04
08:08
1005195711|so   |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 
24-Aug-06 18:05
1005195711|so   |3|00|Application, main: P/N=3150-11069-322
1005195711|app1 |4|00|Initial log entry.
1005195711|app1 |3|00|DNS resolver server is '192.168.15.10'
1005195711|app1 |3|00|DNS resolver search domain is ''
1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash 
e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= 
tn=CircaIP
1005195712|so   |3|00|Link status is Net up Speed 100 full Duplex, PC
down.
1005195722|cfg  |3|00|Beginning to provision phone
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from 
'192.168.15.52'
1005195722|cfg  |3|00|Image bootrom.ld has not changed
1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 
(addr 1 of 1)
1005195722|cfg  |3|00|Downloaded bootROM is identical to Current version

3.2.2
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from

'192.168.15.52'
1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on 
attempt 1 (addr 1 of 1)
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from 
'192.168.15.52'
1005195724|cfg  |3|00|Image sip.ld has not changed
1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 
1 of 1)
1005195724|cfg  |3|00|Downloaded application image is identical to 
current version
1005195724|cfg  |3|00|Phone successfully provisioned
1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
1005195755|app1 |4|00|Loaded application sip.ld successfully, errors
0x0.
1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55
2006


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Re: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

[Ioan Indreias]

Maybe you could use something like:

exten = boss_ext,1,GotoIf($[${CALLERID(number)}=secretary_ext]?boss:secretary)
exten = boss_ext,n(boss),Dial(SIP/boss_ext)
exten = boss_ext,n(secretary),Dial(SIP/secretary_ext)


## nini @ www.modulo.ro ##



Jonathan k. Creasy wrote:
Why don't you just give the secretary the boss' REAL extension and give a different extension to the world that just rings the secretary? 


-jonathan

  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday, January 26, 2007 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Only secretary can call the boss, all others
only reach the secretary when dial the boss extension

Dear all,

How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.

I've tried the following, but it doesn't work:

exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
exten = _boss_extension,1,Dial(SIP/secretary_extension)

This doesn't work because when the secretary tries to transfer the call
to the boss (using her phone's transfer key, not #), one REFER SIP
message is sent back to the caller's phone providing him the new address
for whom the next INVITE should be sent. That INVITE is sent, but when
reaches Asterisk, that INVITE matches this line:

exten = _boss_extension,1,Dial(SIP/secretary_extension)

and not this one:

exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)



Any ideas of how may I solve this issue?
Regards,
Ricardo.
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--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007
11:11 AM




  


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Re: [asterisk-users] Polycom Provistioning Issue

[William M. Conlon]

Looks like the network time server isn't provisioned.

--
Bill
1005195752|app1 |4|00|Could not load time  from 0.0.0.0(0.0.0.0). 

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[asterisk-users] X100P - zttools says red status

[Charlie Grosvenor]

I have an X100P which I have set up as per the guidelines:

 

http://www.x100p.com/support/doc/quick_start_fxo.php

 

The card is recognized by the system:

 

Zaptel Version: 1.4.0

Echo Canceller: MG2

Configuration

==

 

 

Channel map:

 

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

 

1 channels configured.

 

However when I run zttools it says its status is red, which my
understanding of is that it has not detected the line. I am in the uk
and using a standard BT line (with ADSL). Any suggestions?

 

Thanks

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Re: [asterisk-users] Polycom Provistioning Issue

[Jason Walker]

Fixed that issue but it does not change the error
0126204105|cfg  |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 
1 of 1)
0126204105|cfg  |3|00|Downloaded application image is identical to 
current version

0126204105|cfg  |3|00|Phone successfully provisioned
0126204136|app1 |4|00|Loaded application sip.ld successfully, errors 0x0.
0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36 2007

William M. Conlon wrote:

Looks like the network time server isn't provisioned.

--
Bill
1005195752|app1 |4|00|Could not load time  from 0.0.0.0(0.0.0.0). 


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Re: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

[Time Bandit]

How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.

the keyword is context

boss extension : 4321
secretary exten : 4322
in sip.conf for the secretary config, put her phone in the context
secretary-context
for other callers (PSTN lines, other office exten, etc) put them in
context normal-people-context

[normal-people-context]
exten = 4321,1,Dial(SIP/4322)

[secretary-context]
exten = 4321,1,Dial(SIP/4321)

like this, when someone dials 4321, they will reach his secretary,
except when the secretary dials it, she will reach him.

This is just an example written from the top of my head on a friday
afternoon so it is not tested, etc  :)

hth
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RE: [asterisk-users] Polycom Provistioning Issue

[Darryl Dunkin]

Be sure that your mac.cfg file is pointing to a valid configuration
file, I believe the 0x1 error is a missing file error. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:49
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Polycom Provistioning Issue

Fixed that issue but it does not change the error
0126204105|cfg  |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 
1 of 1)
0126204105|cfg  |3|00|Downloaded application image is identical to 
current version
0126204105|cfg  |3|00|Phone successfully provisioned
0126204136|app1 |4|00|Loaded application sip.ld successfully, errors
0x0.
0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36
2007

William M. Conlon wrote:
 Looks like the network time server isn't provisioned.

 --
 Bill
 1005195752|app1 |4|00|Could not load time  from 0.0.0.0(0.0.0.0). 

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Re: [asterisk-users] Polycom Provistioning Issue

[Jason Walker]

?xml version=1.0 standalone=yes?
!-- Default Master SIP Configuration File--
!-- Edit and rename this file to Ethernet-address.cfg for each phone.--
!-- $Revision: 1.14 $  $Date: 2005/07/27 18:43:30 $ --
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=jason.cfg, sip.cfg 
MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= 
CONTACTS_DIRECTORY=/


is my mac IP

Darryl Dunkin wrote:

This is typically an error in one of your config files, either
0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look
like?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue

 From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,

it errors out  with a 0x1 error

Any Ideas?


1005195711|so   |4|00|-- Initial log entry --
1005195711|so   |4|00|+++ Note that bootrom log times are in GMT +++
1005195711|hw   |4|00|Initial log entry.
1005195711|wdog |4|00|Initial log entry
1005195711|cfg  |4|00|Initial log entry
1005195711|copy |3|00|Initial log entry
1005195711|cdp  |4|00|Initial log entry
1005195711|cdp  |5|00|CDP is DISABLED.
1005195711|cdp  |5|00|802.1Q/VLAN tagging is DISABLED.
1005195711|so   |3|00|Platform: Model=SoundPoint IP 501, 
Assembly=2345-11500-040 Rev=A

1005195711|so   |3|00|Platform: Board=2345-11500-040 A
1005195711|so   |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, 
Subnet Mask=255.255.255.0

1005195711|so   |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04
08:08
1005195711|so   |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 
24-Aug-06 18:05

1005195711|so   |3|00|Application, main: P/N=3150-11069-322
1005195711|app1 |4|00|Initial log entry.
1005195711|app1 |3|00|DNS resolver server is '192.168.15.10'
1005195711|app1 |3|00|DNS resolver search domain is ''
1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash 
e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= 
tn=CircaIP

1005195712|so   |3|00|Link status is Net up Speed 100 full Duplex, PC
down.
1005195722|cfg  |3|00|Beginning to provision phone
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from 
'192.168.15.52'

1005195722|cfg  |3|00|Image bootrom.ld has not changed
1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 
(addr 1 of 1)

1005195722|cfg  |3|00|Downloaded bootROM is identical to Current version

3.2.2
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg' from

'192.168.15.52'
1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on 
attempt 1 (addr 1 of 1)
1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from 
'192.168.15.52'

1005195724|cfg  |3|00|Image sip.ld has not changed
1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 
1 of 1)
1005195724|cfg  |3|00|Downloaded application image is identical to 
current version

1005195724|cfg  |3|00|Phone successfully provisioned
1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
1005195755|app1 |4|00|Loaded application sip.ld successfully, errors
0x0.
1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55
2006


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Re: [asterisk-users] 1.4 sounds long space before and after prompt

[Ry]

I noticed the same problem as well. I will see if I the old sound files
corrects the problem, or if it's actually a timing problem.
I have to say, I like the old sounds better, they sounded softer.
-Ry

On 12/17/06, Gil Kloepfer [EMAIL PROTECTED] wrote:


Is anyone else finding in the new audio files that the longer space
at the beginning and end of the files tends to be extremely irritating?
An excellent example is when going into voicemail and Allison says how
many messages you have, the space between the files is annoyingly long:

   you have .. four .. old .. messages

..and..

  first .. message .. received . July . twenty ..
second

Under the old sound files, this continuity was still a little long,
but workable.  The new sound files make these positively sound like
a computer playing individual files rather than a continuous sentence.

If I release these sound files as they are to my users, they are going
to revolt.  They already complain about the old Octel VM system prompts
being played back too slowly and these are much slower than that.

I mentioned this a while back when the new sounds were in beta, but
haven't seen anything more about it.  So either this says something
about my and my users' level of patience, I'm missing something
that changed between 1.2 and 1.4 that could fix this, or the
focus has been on lower-level issues with 1.4 than on the sound files.

With the new higher-quality sound files, I could manually edit all
the offending files (there are lots of them) and correct what I perceive
to be a problem.  However, if this is a common enough complaint, maybe
others would want to help as well, and we could get the fixed files
put back into core Asterisk.

Note that this doesn't appear to be a problem with the speed of the
sound files as some others have experienced.  The tempo is probably okay,
and the pitch is fine.  It's the spacing between files that's the issue
I'm talking about.

Thanks in advance for any feedback.

---
Gil Kloepfer
[EMAIL PROTECTED]
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[asterisk-users] Nobody there, continuing...

[Alex Robar]

Hi all,

Running Asterisk 1.2.12 (a bit out dated, but it was fully operational until
a few days ago), I'm seeing the following message in my logs, repeated
literally millions of times:

channel.c: Nobody there, continuing…

We've started to see some odd behavior (incoming callers can hear us, we
can't hear them, we can't dial out, etc). I read that this error might
possibly be related to not setting rtptimeout, but I've set this and the
issue persists. The symptoms seem very familiar to the types of issues we
see when the internet goes down (call routing seems to get all screwy), but
the connection appears to be fully operational when the symptoms appear. A
reboot fixes the issues for about 3/4 of a day, but then they start
happening again. Does anybody out there have any clue as to the meaning of
the nobody there message is?

Thanks,
Alex Robar
--
Alex Robar
[EMAIL PROTECTED]
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[asterisk-users] Show call coming back from Call Parking

[Asterisk User List]

Our operator has asked if it is possible that when a call times out in
the call parking and comes back to her, if there is someway to show that
call has come back from parking.  I have looked all over the
documentation and have come up with nothing so far.

All I see when a call times out is:
-- Stopped music on hold on Zap/25-1
  == Timeout for Zap/25-1 parked on 702. Returning to
park-dial,SIP/214,1
-- Executing Dial(Zap/25-1, SIP/214||t) in new stack
-- Called 214
-- SIP/214-09086ff8 is ringing

It appears that the park-dial is a context that Asterisk autogenerates
so there is nothing I can do in that context.

Has anyone else found a way to show that this a call returning and not a
new call coming in?

Thanks
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[asterisk-users] Sample Config.

[Jonson Player]

Hello,
I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to
configure voice part on it. I cannot get it if I can use like peer for my
asterisk. Please help me with some tips.
Thank you guys.

Regards,
Jonson.
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Re: [asterisk-users] convert URI string to lowercase

[Ioan Indreias]

Hello,

Maybe using app_backticks will solve your problem.
I use it to call a script and saved the result into an Asterisk variable.

http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks

Regards,
## nini @ www.modulo.ro ##

Pavel Jezek wrote:
any idea, how to do something like this, but in correct/functional 
form?  ;-)


Set(foo=System(echo ${EXTEN} | tr [:upper:] [:lower:]))

${EXTEN} is SomeStrinG
${foo} output should bee somestring
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RE: [asterisk-users] Polycom Provistioning Issue

[Darryl Dunkin]

Looks alright there. The next config to check is where it loads your
'jason.cfg', any errors will be in your app logfile (as opposed to the
boot one you pasted).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 13:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Provistioning Issue

?xml version=1.0 standalone=yes?
!-- Default Master SIP Configuration File--
!-- Edit and rename this file to Ethernet-address.cfg for each
phone.--
!-- $Revision: 1.14 $  $Date: 2005/07/27 18:43:30 $ --
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=jason.cfg, sip.cfg 
MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= 
CONTACTS_DIRECTORY=/

is my mac IP

Darryl Dunkin wrote:
 This is typically an error in one of your config files, either
 0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look
 like?

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jason
 Walker
 Sent: Friday, January 26, 2007 12:10
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Polycom Provistioning Issue

  From what I know this log show everything working Perfect.
 Then it goes to the Welcome screen then after a long time of
processing,

 it errors out  with a 0x1 error

 Any Ideas?


 1005195711|so   |4|00|-- Initial log entry --
 1005195711|so   |4|00|+++ Note that bootrom log times are in GMT +++
 1005195711|hw   |4|00|Initial log entry.
 1005195711|wdog |4|00|Initial log entry
 1005195711|cfg  |4|00|Initial log entry
 1005195711|copy |3|00|Initial log entry
 1005195711|cdp  |4|00|Initial log entry
 1005195711|cdp  |5|00|CDP is DISABLED.
 1005195711|cdp  |5|00|802.1Q/VLAN tagging is DISABLED.
 1005195711|so   |3|00|Platform: Model=SoundPoint IP 501, 
 Assembly=2345-11500-040 Rev=A
 1005195711|so   |3|00|Platform: Board=2345-11500-040 A
 1005195711|so   |3|00|Platform: MAC=0004f2023ecc, IP=192.168.15.55, 
 Subnet Mask=255.255.255.0
 1005195711|so   |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04
 08:08
 1005195711|so   |3|00|Application, main: Label=BOOT,
Version=3.2.2.0019 
 24-Aug-06 18:05
 1005195711|so   |3|00|Application, main: P/N=3150-11069-322
 1005195711|app1 |4|00|Initial log entry.
 1005195711|app1 |3|00|DNS resolver server is '192.168.15.10'
 1005195711|app1 |3|00|DNS resolver search domain is ''
 1005195711|app1 |3|00|Bootline: eim(0,0)bootHost:flash 
 e=192.168.15.55:ff00 h=192.168.15.52 g=192.168.15.10 u=1 pw= 
 tn=CircaIP
 1005195712|so   |3|00|Link status is Net up Speed 100 full Duplex, PC
 down.
 1005195722|cfg  |3|00|Beginning to provision phone
 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/bootrom.ld' from 
 '192.168.15.52'
 1005195722|cfg  |3|00|Image bootrom.ld has not changed
 1005195722|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 
 (addr 1 of 1)
 1005195722|cfg  |3|00|Downloaded bootROM is identical to Current
version

 3.2.2
 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/0004f2023ecc.cfg'
from

 '192.168.15.52'
 1005195722|copy |3|00|Download of '0004f2023ecc.cfg' succeeded on 
 attempt 1 (addr 1 of 1)
 1005195722|copy |3|00|'ftp://1:[EMAIL PROTECTED]/sip.ld' from 
 '192.168.15.52'
 1005195724|cfg  |3|00|Image sip.ld has not changed
 1005195724|copy |3|00|Download of 'sip.ld' succeeded on attempt 1
(addr 
 1 of 1)
 1005195724|cfg  |3|00|Downloaded application image is identical to 
 current version
 1005195724|cfg  |3|00|Phone successfully provisioned
 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
 1005195755|app1 |4|00|Loaded application sip.ld successfully, errors
 0x0.
 1005195755|app1 |6|00|Uploading boot log, time is THU OCT 05 19:57:55
 2006


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[asterisk-users] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms

[Naija Man]

Hello,

We have an asterisk system with about 40 cisco 7940/7960 phones and a few
linksys SPA941. I recently analyzed our network and discovered that the rtp
packet size from the cisco phones is 10ms. We want to change the rtp packet
size of the Cisco phones from 10ms to 20ms. I know how to do this on linksys
phones and sipura ATAs but I cannot figure out how on the 7940/7960s. Is
this possible? Does anyone have suggestions as to how I can do achieve this?
Any tip or help will be appreciated.

Codec: ULAW
SIP firmware: 8.2

Thanks.

Buki
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Re: [asterisk-users] Show call coming back from Call Parking

[Eric \ManxPower\ Wieling]

Asterisk User List wrote:

Our operator has asked if it is possible that when a call times out in
the call parking and comes back to her, if there is someway to show that
call has come back from parking.  I have looked all over the
documentation and have come up with nothing so far.

All I see when a call times out is:
-- Stopped music on hold on Zap/25-1

  == Timeout for Zap/25-1 parked on 702. Returning to
park-dial,SIP/214,1
-- Executing Dial(Zap/25-1, SIP/214||t) in new stack
-- Called 214
-- SIP/214-09086ff8 is ringing

It appears that the park-dial is a context that Asterisk autogenerates
so there is nothing I can do in that context.

Has anyone else found a way to show that this a call returning and not a
new call coming in?


[park-dial]
exten = _.,1,SetCIDName(Parking Timeout)
exten = _.,2,SetVar(__ALERT_INFO=Triplet)
exten = _.,3,Goto(extensions,3500,1)


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Re: [asterisk-users] NTL Hangup

[Leo Ann Boon]

Kyle Gordon wrote:


fxsks=1 #X100P

Is your line truly a kwelstart line? try fxsls

SNIP
busydetect=yes

You may need to add these 2 values to help the busydetect
busycount=3
busypattern=375,375

busypattern tells asterisk how your busy tone sounds like, in UK it 
should be 400Hz 0.375s ON and 0.375s OFF. The busycount tells asterisk 
how many consecutive cycles it must detect before dropping the line. 
You'll have to determine the best value for your setup, by trial and 
error. Too low - you might get premature hangup, too high - you'll have 
to wait for a long time for the line to hangup. A value of 3 will cause 
Asterisk to hang up in about 2.1s.



SNIP
switchtype=national

This is not needed for analog lines.

signalling=fxs_ks

Change to fxs_ls to match zaptel.conf

SNIP
I don't know the tone plan for NTL. They seem to use a different tone 
for hanging up from BT, but I'm not sure how to go about implementing 
any changes in the configs to reflect it.

If it's different you'll need to modify zonedata.c in the zaptel directory.

Leo.
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Re: [asterisk-users] X100P - zttools says red status

[Tzafrir Cohen]

On Fri, Jan 26, 2007 at 08:42:36PM -, Charlie Grosvenor wrote:
 I have an X100P which I have set up as per the guidelines:
 
 http://www.x100p.com/support/doc/quick_start_fxo.php
 
 The card is recognized by the system:
 
 Zaptel Version: 1.4.0
 
 Echo Canceller: MG2
 
 Configuration
 
 ==
 
  
 
  
 
 Channel map:
 
  
 
 Channel 01: FXS Kewlstart (Default) (Slaves: 01)
 
  
 
 1 channels configured.
 
  
 
 However when I run zttools it says its status is red, which my
 understanding of is that it has not detected the line. I am in the uk
 and using a standard BT line (with ADSL). Any suggestions?

Is the line plugged in? Can you connect a standard phone to the same
line?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] NTL Hangup

[Tzafrir Cohen]

On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote:
 Kyle Gordon wrote:
 
 fxsks=1 #X100P
 Is your line truly a kwelstart line? try fxsls

And if the line is ls, indeed, what harm is there in setting it up as
ks?

Consider, e.g.
http://svn.digium.com/svn/asterisk-gui/trunk/tools/zapscan.c

 SNIP
 busydetect=yes
 You may need to add these 2 values to help the busydetect
 busycount=3
 busypattern=375,375

this should have been progzone=uk , only it turns out that the UK
progzone actually sets it to 400.

I'd like to ask again: 
where are you using specific progzones and buzypatterns successfully?
Those magic values should be better documented.

 
 busypattern tells asterisk how your busy tone sounds like, in UK it 
 should be 400Hz 0.375s ON and 0.375s OFF. The busycount tells asterisk 
 how many consecutive cycles it must detect before dropping the line. 
 You'll have to determine the best value for your setup, by trial and 
 error. Too low - you might get premature hangup, too high - you'll have 
 to wait for a long time for the line to hangup. A value of 3 will cause 
 Asterisk to hang up in about 2.1s.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] NTL Hangup

[Leo Ann Boon]

Tzafrir Cohen wrote:

On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote:
  

Kyle Gordon wrote:


fxsks=1 #X100P
  

Is your line truly a kwelstart line? try fxsls



And if the line is ls, indeed, what harm is there in setting it up as
ks?
  
I understand ks is ls with a wink start. In some cases, use ks on a ls 
line will cause bizarre problems.

Consider, e.g.
http://svn.digium.com/svn/asterisk-gui/trunk/tools/zapscan.c

  

SNIP
busydetect=yes
  

You may need to add these 2 values to help the busydetect
busycount=3
busypattern=375,375



this should have been progzone=uk , only it turns out that the UK
progzone actually sets it to 400.
  
The progzone uk is actually correct, 400Hz, 375ms ON and 375ms OFF. But 
,I believe it's not actually used in the busy detector. See this 
explanation from Steve Davis on why busypattern was added to zapata.conf

http://bugs2.digium.com/print_bug_page.php?bug_id=4830
I'd like to ask again: 
where are you using specific progzones and buzypatterns successfully?

Those magic values should be better documented.
  
I agree with you, this is voodoo magic :). I'd only figured out for 
myself by trial and error.



Leo
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RE: [asterisk-users] X100P - zttools says red status

[Charlie Grosvenor]

Yes the line is connected, a standard phone works fine when connected to
the line.

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 26 January 2007 23:45
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] X100P - zttools says red status

On Fri, Jan 26, 2007 at 08:42:36PM -, Charlie Grosvenor wrote:
 I have an X100P which I have set up as per the guidelines:
 
 http://www.x100p.com/support/doc/quick_start_fxo.php
 
 The card is recognized by the system:
 
 Zaptel Version: 1.4.0
 
 Echo Canceller: MG2
 
 Configuration
 
 ==
 
  
 
  
 
 Channel map:
 
  
 
 Channel 01: FXS Kewlstart (Default) (Slaves: 01)
 
  
 
 1 channels configured.
 
  
 
 However when I run zttools it says its status is red, which my
 understanding of is that it has not detected the line. I am in the uk
 and using a standard BT line (with ADSL). Any suggestions?

Is the line plugged in? Can you connect a standard phone to the same
line?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] X100P - zttools says red status

[Leo Ann Boon]

Charlie Grosvenor wrote:

Yes the line is connected, a standard phone works fine when connected to
the line.
  
There're 2 ports on the card. Which port are you using? One of the ports 
is for connecting another phone in parallel to the card.


Leo
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Re: [asterisk-users] Asterisk 1.4 Polycom buddy status

[James Fromm]

Olle E Johansson wrote:


26 jan 2007 kl. 16.31 skrev James Fromm:


Olle E Johansson wrote:

24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:

James Fromm wrote:

The behavior we see is that the SIP interface in the queue will 
sometimes not release from the in-use state.  Connecting to the 
interface from another SIP device and immediately hanging up will 
clear the state.
The phones in question are configured with one line that will 
except only one call.  The device itself does not think it is 
in-use because it will accept another call.  Something in the SIP 
channel driver is not clearing the state when a call is completed.
There is definitely no correlation between this and Asterisk 
restarting.  In fact, if a device is 'stuck' on in-use, restarting 
Asterisk will clear the state.
I've been working on this for a week now.  It only started for us 
because I just implemented the call-limit option in the sip.conf in 
Asterisk for the devices.  See my posts with subject 'Queue and 
Interface time out'.


I believe there is/was a bug relating to call-limit.  Buddy Watch 
doesn't work if you use call-limit and if a call from a queue is 
transfered, the call-limit is not released until the original call 
is terminated.  I do not know if these issues have been fixed or not.
Again, a relation to call transfer. I think the bug is that we don't 
handle call-limits properly during a call transfer. That needs

to be verified and fixed.


There may be, but transfers are not the cause of the issue I describe. 
SIP interfaces that are members of a Queue, will erratically not be 
released from 'in-use' when a call is completed.  I have tested with 
both caller terminated and agent terminated calls and both will cause 
this behavior.  It happens on approximately 20% of all calls the queue 
members receive.  Dialing the SIP device with another device will 
immediately free the status.


I wonder if this only happens on calls sent to the SIP device by the 
Queue application.  I will test that today.


If you are using chan_agent as a proxy channel, check if that changes 
things.




We don't have agents defined so I don't think chan_agent applies.  The 
Queue's members are assigned through the management port from an 
application running on the the agent's PC.  I think the Queue 
application loses sync to the SIP channel driver's information 
containing the state of the SIP interfaces.


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Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!!

[kjcsb]

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

Sent: Wednesday, January 24, 2007 8:24 AM
Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends... 
bigheadache!!






hi folks,

I am using asterisk 1.2.13 (debian etch).

My customer's sip accounts are stored in realtime sipusers.

I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes

Each account has nat=yes

Now, I have lot of problems.

for example, when I change the 'secret'  field of a user in the database, 
it

doesn't
get reflected in Asterisk, who is still expecting the old password.

As far as I know when rtcachefriends=yes database changes are unavailable 
to Asterisk until a reload is performed.


Cameron 


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Re: [asterisk-users] Sample Config.

[Token PBX]

Hi!



I don't understand  what you mean by : „configure voice part on it, but I
can give general guidelines:



First you setup SPA3000 web UI:

1) Line1 Tab:



Sip settings:

  SIP port : 5060



Proxy and Registration:

  Proxy: Asterisk IP



Subscriber Information:

  Display Name: FXS_username

  Password: FXS password

  User ID: FXS_username



2) PSTN Line Tab:



SIP Settings:

  SIP port: 5061



Proxy and Registration:

  Proxy: Asterisk IP



Subscriber Information:

  Display Name: FXO_username

  Password: FXO_password

  User ID: FXO_username



Dial Plans:

  Dial Plan 1: (S0:[EMAIL PROTECTED] IP:5060)(may be any other dial plan)



VoIP-To-PSTN Gateway Setup:

  VoIP-To-PSTN Gateway Enable: Yes

  Line 1 VoIP Caller DP: 1 (or any other setup like Dial Plan 1)



VoIP Users and Passwords (HTTP Authentication)

  VoIP User 1 Auth ID: asterisk

  VoIP User 1 DP: 1(same as above)



PSTN-To-VoIP Gateway Setup:

  PSTN-To-VoIP Gateway Enable: Yes





Then Asterisk sip.conf:



[ FXO_username]

disallow=all

allow=alaw

type=friend

fromuser= FXO_username

username= FXO_username

secret= FXO_password

host=dynamic

dtmfmode=rfc2833

canreinvite=no

qualify=1000

context=incoming

port=5061



[FXS_username]

disallow=all

allow=alaw

type=friend

username= FXS_username

secret= FXS_password

host=dynamic

dtmfmode=rfc2833

canreinvite=no

qualify=1000

context=outgoing

Best regards
Mihaela MJ


On 1/26/07, Jonson Player [EMAIL PROTECTED] wrote:


Hello,
I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to
configure voice part on it. I cannot get it if I can use like peer for my
asterisk. Please help me with some tips.
Thank you guys.

Regards,
Jonson.

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Re: [asterisk-users] Asterisk 1.4 Polycom buddy status

[Anthony Rodgers]

Hi there,

We traced this issue to transfers (see http://bugs.digium.com/ 
view.php?id=8848). On Monday, I will be attaching the various debugs  
to the case as requested.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp



On 26-Jan-07, at 5:16 PM, James Fromm wrote:




Olle E Johansson wrote:

 26 jan 2007 kl. 16.31 skrev James Fromm:

 Olle E Johansson wrote:
 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
 James Fromm wrote:

 The behavior we see is that the SIP interface in the queue will
 sometimes not release from the in-use state.  Connecting to the
 interface from another SIP device and immediately hanging up  
will

 clear the state.
 The phones in question are configured with one line that will
 except only one call.  The device itself does not think it is
 in-use because it will accept another call.  Something in the  
SIP
 channel driver is not clearing the state when a call is  
completed.

 There is definitely no correlation between this and Asterisk
 restarting.  In fact, if a device is 'stuck' on in-use,  
restarting

 Asterisk will clear the state.
 I've been working on this for a week now.  It only started  
for us
 because I just implemented the call-limit option in the  
sip.conf in

 Asterisk for the devices.  See my posts with subject 'Queue and
 Interface time out'.

 I believe there is/was a bug relating to call-limit.  Buddy Watch
 doesn't work if you use call-limit and if a call from a queue is
 transfered, the call-limit is not released until the original  
call
 is terminated.  I do not know if these issues have been fixed  
or not.
 Again, a relation to call transfer. I think the bug is that we  
don't

 handle call-limits properly during a call transfer. That needs
 to be verified and fixed.

 There may be, but transfers are not the cause of the issue I  
describe.

 SIP interfaces that are members of a Queue, will erratically not be
 released from 'in-use' when a call is completed.  I have tested  
with
 both caller terminated and agent terminated calls and both will  
cause
 this behavior.  It happens on approximately 20% of all calls the  
queue

 members receive.  Dialing the SIP device with another device will
 immediately free the status.

 I wonder if this only happens on calls sent to the SIP device by  
the

 Queue application.  I will test that today.

 If you are using chan_agent as a proxy channel, check if that  
changes

 things.


We don't have agents defined so I don't think chan_agent applies.  The
Queue's members are assigned through the management port from an
application running on the the agent's PC.  I think the Queue
application loses sync to the SIP channel driver's information
containing the state of the SIP interfaces.

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[asterisk-users] IP-to-IP dial: no answer or no listener?

[Yuan LIU]

Dial(SIP/[EMAIL PROTECTED]) will ring forever even if no application is 
listening.  How can Asterisk tell if the user is not answering or simply not 
having SIP?


Yuan Liu


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[asterisk-users] Does X100P decode caller ID?

[Yuan LIU]

The SM56 MODEM manual says it does.  But when used with zaptel 1.2.12, 
nothing shows up.


Yuan Liu


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RE: [asterisk-users] Does X100P decode caller ID?

[Yuan LIU]

From: Yuan LIU [EMAIL PROTECTED]

The SM56 MODEM manual says it does.  But when used with zaptel 1.2.12, 
nothing shows up.


Debug level 6 (Asterisk 1.4.0) only shows:
[Jan 26 22:56:46] ERROR[25603]: callerid.c:564 callerid_feed: fsk_serie made 
mylen  0 (-14)
[Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6389 ss_thread: CallerID feed 
failed: Success
[Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6489 ss_thread: CallerID 
returned with error on channel 'Zap/1-1'


Any idea?


Yuan Liu



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[asterisk-users] Digium AIX demo nogo (was: NAT solutions)

[Yuan LIU]

From: Tim Panton [EMAIL PROTECTED]
Thanks for this information.  Does this mean two IAX boxes can talk  
behind their respective NAT's (without any server sitting in voice  path)? 
 I'm imagining this:


Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2

If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.


Yes, with 1 proviso - one end needs a known IP address and a port map
for udp 4569 in the router. The other can simply register to it with  zero
router config.

Tim Panton

www.mexuar.net
www.westhawk.co.uk/


Unrelated to dual firewalls - I just tried Asterisk Demo included in sample 
configs.  Priority 2 in extension 500 is

 Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
But the result is nogo:
   -- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, 
IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack

   -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
[Jan 26 22:58:13] NOTICE[25383]: chan_iax2.c:2686 __auto_congest: 
Auto-congesting call due to slow response

   -- IAX2/216.207.245.8:4569-1 is circuit-busy
   -- Hungup 'IAX2/216.207.245.8:4569-1'
 == Everyone is busy/congested at this time (1:0/1/0)

I am behind a NAT that one SIP provider has no problem penetrating (no port 
forwarding).  I then opened 4569 to my Asterisk.  Still no go.


Thank you for suggestions.

Yuan Liu


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Re: [asterisk-users] NAT solutions

[Brad Templeton]

On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
 Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
 
 If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.

While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of the NATs to
open ports, or if the asterisk servers know the address of their
partner and thus can keep the NAT open by sending keep-alives.
 
 The way Jeff Pulver puts it, ICE has conquered the world :-)  Would love 
 to learn more.

ICE is a methodology.  You list every way you might be reached
(LAN, external addresses and addresses of outside relays) and the
other endpoint tries every way it can, ranked in order of quality,
and picks the best one.   So if you're both on the same LAN it will
see that and use it.  If you can't reach one another except through
a relay it identifies that and uses a relay.  If, of course, you have
a willing relay.

(Skype solved that last problem :-)
 
 Is this the concept of STUN?  Does this also create latency (by adding an 
 additional leg in the route), packet loss, even jitter?
STUN is something else.  Using a relay does indeed increase latency
(and thus echo) and may increase jitter and packet loss, though latency
is the big issue.
 
 I should have used FWD as an example.  One can't say it uses proprietary 
 clients.  Does it stay away from voice path?

It provides a relay if one is needed.  I don't know about today but
they started using jasomi boxes sold to deal with this question.
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Re: [asterisk-users] NAT solutions

[Brad Templeton]

On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
 For a remote phone, not on the same network as the Asterisk
 box (in which event the NAT worries are different) you definitely
 want to use the same protocol for the phone as for your
 term/orig provider.   Otherwise you will be forced to hairpin
 your audio through your asterisk server, adding latency and
 wasting bandwidth and cpu for little reason.
 
 Unless you are monitoring calls, want full CDR  etc,
 then that's what you want anyway.

CDR are not affected by how the audio flows.  Monitoring
calls does require hairpin of the audio.  Most people who
are not call centers do not wish to monitor all calls or
even more than few calls.  (In fact in many states it is
illegal unless you inform the other party, mostly limiting
it to call center use.)

If you had a call center * server in the USA hairpinning
a call between India and the UK it would be really dumb,
but even over shorter links it's dumb.
 
 I agree. Single SIP phones can usually be got to work behind
 a reasonable NAT router.

And with some work could be made to work without special
config with all but the rarest NATs.  Hopefully in 1.6.
 
 For a single phone - you are quite right. For multiple phones,
 I'm not sure I agree - multiple SIP phones behind a NAT router
 is going to require some extensive config , or a SIP proxy in the  
 router.

Not really, other than the issue of NATS that won't hairpin
between the phones.

I have this situation, and our 2nd home I have 2 phones, on
the * server at my main home.  While I have linux computers at
the 2nd home, it would be silly to put up a * server for the
two phones if they can work through the NAT.  It's not a big
deal to tell the WRT54G I have to forward two ports to the 2
phones (well 3 if you include the wifi phone).  There is
no need at the 2nd house for intercom, so I would not put in
a local server just for that.

However, it does mean the remote location can't have SIP phones
without things like STUN.

 Ah, but it isn't just asterisk you have to change - it is
 all the SIP implementations and all the routers :-)

STUN is quite common in SIP phones, in fact the only major modern
ones to not do it seem to be the Ciscos, though I have not
tried the 8.0 firmware on them.
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RE: [asterisk-users] Does X100P decode caller ID?

[Yuan LIU]

From: Yuan LIU [EMAIL PROTECTED]

Debug level 6 (Asterisk 1.4.0) only shows:
[Jan 26 22:56:46] ERROR[25603]: callerid.c:564 callerid_feed: fsk_serie 
made mylen  0 (-14)
[Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6389 ss_thread: CallerID feed 
failed: Success
[Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6489 ss_thread: CallerID 
returned with error on channel 'Zap/1-1'


A little googling made me realize that Asterisk demo may not be the best 
application to look for caller ID because it tries to pick up at first ring. 
 So I zapped demo context with a plain one.  This time, no more failed 
success.  But Asterisk only receives

  New User,
no matter which caller calls. (Callers can be correctly identified from 
other devices.)


The machine doesn't have sound card, so experimenting with rxgain would be 
difficult - but guess my best bet is to find a way to do this.


Yuan Liu


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