Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

First off...  please cancel previous amplification request.  I have  
implemented your ideas with the same errored result.

I am not sure that we are not making it thru authentication.  From my  
digging and comparing packet dumps comparing the soft phone to asterisk  
they have identical transactions through  the ACK reply (the last one  
on the debug below).  The softphone seems to be authenticated after the  
ACK.  I am a newbie to debugging this stuff. I just want to get it  
working.

Thanks everyone in advance for your help.  I am certainly very very  
happy to try anything.

Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice  
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

End result...
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
;tag=as545ccba3'

SIP debug...
-- Executing Dial("SIP/6050-132b",  
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest  
username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 
129dd4fb5f97ec47"
Contact: 6050 
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 ;tag=7e2776985d5a0891o0
To: ;tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0
 to 64.4.192.110:5060
com*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST  
realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" ;tag=as545ccba3
To: ;tag=SD38rq699-
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 147.135.8.128:5060
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
;tag=as545ccba3'


On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing

Re: [Asterisk-Users] Forwarded call flag

[Tom Samplonius]

On Tue, 8 Mar 2005 13:36:39 -0700, Dr. Matthew Roller
<[EMAIL PROTECTED]> wrote:
> When I forward my PSTN phone(Qwest) to my cellphone and someone calls
> it, my cellphone(AT&T) shows an arrow next to the caller id showing it
> is a forwarded call, is there any way to set that forward flag when
> forwarding a phone call through asterisk?

  It is possible using PRI signalling (ni2 for sure).  When running
q931 debugging on a PRI gateway, I can see that incoming calls that
have been forwarded actually have the redirecting number listed, and
the type of redirection (always, na/busy, etc).  I really doubt that
libpri has the capability to build these kinds of messages, let alone
an API to set the fields.  And then try to find a PRI provider won't
strip the messages (called GTD, if you talk to them).

  However, Cisco gateways running SIP can do this.  They are actually
able to set the redirecting number in the INVITE, and pass it to
another gateway, which can regenerate it back into a PRI message. 
Asterisk doesn't understand redirecting numbers in SIP INVITES though,
so this only works with gateway to gateway calls.

Tom
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RE: [Asterisk-Users] Another Newbie Question

[Jim Van Meggelen]

Callum McGillivray wrote:
> Hey all,
> 
> My apologies if this sounds blindingly obvious, but am I correct in
> saying that I can use Asterisk to connect two extensions and make
> calls between them without needing an actual telephone line at all ?  
> 
> As I said, probably blindingly obvious… but my techies have gone home
> for the evening and I was looking for an answer before I left. 

You could do that with two tin cans and a string! ;-P

In all seriousness, the answer to your question is: yes, Asterisk can
do that, and a whole lot more. 

Cheers,


--
Jim Van Meggelen
[EMAIL PROTECTED]

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 07/03/2005
 

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RE: [Asterisk-Users] i am missing something!

[Jay Milk]

You'll need canreinvite=no to each sip section in sip.conf, if you want
* to stay in the loop.

> -Original Message-
> From: Adnan Ahmed [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, March 09, 2005 1:14 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] i am missing something!
> 
> 
> Hello ppl,
> At initial level i configure asterisk woth only soft phones 
> ,in which one at windows machine and other is linux i am 
> using windows messenger and linphone respectively both phones 
> registered with asterisk respectively problem is that they 
> bypass asterisk on call when i send request from linphone to 
> messenger request shown on messenger but on asterisk console 
> nothing to and also if i send request from messenger to 
> linphone it doesn't recognized at all my config are:

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[Asterisk-Users] Another Newbie Question

[Callum McGillivray]

Hey all,

 

My apologies if this sounds blindingly obvious, but am I
correct in saying that I can use Asterisk to connect two extensions and make
calls between them without needing an actual telephone line at all ?

 

As I said, probably blindingly obvious… but my techies
have gone home for the evening and I was looking for an answer before I left.

 

Thanks,

 

Callum






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[Asterisk-Users] i am missing something!

[Adnan Ahmed]

Hello ppl,
At initial level i configure asterisk woth only soft phones ,in which
one at windows machine and other is linux i am using windows messenger
and linphone respectively both phones registered with asterisk
respectively problem is that they bypass asterisk on call when i send
request from linphone to messenger request shown on messenger but on
asterisk console nothing to and also if i send request from messenger
to linphone it doesn't recognized at all my config are:
extensions.conf:
[general]
static=yes
writeprotect=no


[sip]
exten => 101,1,Dial(SIP/101,20) >msn
exten => 922,2,Dial(SIP/102,20) ->linphone

sip.conf:
[general]
context=sip 
port=5060   
bindaddr=192.168.0.50   (asterisk server ip)
maxexpirey=3600 
defaultexpirey=120  
disallow=all
allow=ulaw  
allow=alaw
allow=gsm
relaxdtmf=yes   
rtptimeout=60   
rtpholdtimeout=300  
;useragent=Asterisk PBX 
;nat=no 


[911]
username=101
type=friend
callerid=101
context=sip
qualify=no
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
;allow=gsm
defaultip=192.168.0.60


[912]
username=102
type=friend
host=dynamic
dtmfmode=inband
context=sip
disallow=all
allow=alaw
allow=ulaw
;allow=gsm
nat=no
defaultip=192.168.0.51
canreinvite=yes

what i want when asterisk registers it can only make calls otherwise refuse it .
Don't bother with my question.
Thank You.
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Re: [Asterisk-Users] Connect asterisk on classic pbx with T100P card

[Peter Svensson]

On Wed, 9 Mar 2005, bagattin jerome wrote:

> Hi, 
> 
> I try to connect my asterisk box with a classic pbx
> (Siemens).
> I have a T100P E1 card.

The T100P is a T1 card, not an E1. Did you mean to write E100P? 

> modprobe zaptel and wct1xxp ok
> ztcfg -vv  ok
> 
> I can run asterisk and show the channels but the led
> still flash red
> 
> My zaptel.conf :
> 
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> bchan=17-31
> dchan=16

Is the pbx set to provide timing? If not, you need to change to span=1,0,0
Also, check that crc4 is enabled on the pbx as well.

Peter

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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

First, thanks for your help.
I have been changing these to different values but not getting it. 
Could you further amplify your statement...

Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf.
Thanks!
Chris
On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2
In /etc/hosts I have:
147.135.0.128   sip.broadvoice.com
It's the proxy.dca.broadvoice.com server. Hope this helps...
--Luki
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[Asterisk-Users] Connect asterisk on classic pbx with T100P card

[bagattin jerome]

Hi, 

I try to connect my asterisk box with a classic pbx
(Siemens).
I have a T100P E1 card.

modprobe zaptel and wct1xxp ok
ztcfg -vv  ok

I can run asterisk and show the channels but the led
still flash red

My zaptel.conf :

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


My zapata.conf :

;; E1
switchtype = euroisdn
signalling = pri_net
group = 1
context=e1
channel => 1-15,17-31

The pbx T2 in the pbx is configured as a S2M euroisdn
with crc

The cable between asterisk and pbx look like that :


asterisk   1   -4pbx
   2   -5
   4   -1
   5   -2

I have also try simple patch-cable (wire 1 to 1).

what's going wrong ? config ? cable ?
I have no more idea.

Thanks for your help

Jerome







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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Luki]

Chris,

first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.

That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.

I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])

Try changing yours to say "broadvoice" and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:

[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2

In /etc/hosts I have:
147.135.0.128   sip.broadvoice.com

It's the proxy.dca.broadvoice.com server. Hope this helps...

--Luki
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RE: [Asterisk-Users] Broadvoice-like company in Canada?

[Jay Milk]

> -Original Message-
> Is it easy to use Vonage with an * box? Could I order service 
> from them and not use the equip. they send?

Search this list for "vonage".  Softphone's possible, hardphone is not.

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[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

[Chris Nibeck]

I have been going crazy with this also since Sat.
Our server was working perfectly with BV but will now not place calls 
to BV.

Incoming from BV works fine.
I felt sad rebooting it today, it had been running for almost 200 days!
Here is my error message from the console...
Notice I am running today's CVS
Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com 
(pid = 1624)
-- Executing Dial("SIP/6050-5bc9", 
"SIP/[EMAIL PROTECTED]|30") in new stack
-- Called [EMAIL PROTECTED]
Mar  8 23:11:55 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed 
to authenticate on INVITE to '"6050" 
;tag=as20911f6e'

I have tried many versions of sip.conf, here is the current...
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=blah
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
I have tried the different proxies proxy.dca.broadvoice.com, lax, mia, 
and was originally using chi when the system worked.

BV told me Mon that chi is considered a test server that should not be 
used for production, it is expected to go up and down.

My hosts file points to one of the working ones.
I verified my account through a softphone. It works fine to BV.
There is something wrong with the authentication.
Here is the SIP debug...
-- Executing Dial("SIP/6050-019c", 
"SIP/[EMAIL PROTECTED]|30") in new stack
We're at xxx.xxx.xxx.xxx port 16776
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 05:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3501 3501 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 16776 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI>
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: ;tag=SD38ad399-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST 
realm="BroadWorks",algorithm=MD5,nonce="1110346372627"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c
From: "6050" ;tag=as292b9469
To: ;tag=SD38ad399-
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 147.135.8.128:5060
Mar  8 23:35:15 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed 
to authenticate on INVITE to '"6050" 
;tag=as292b9469'

TIA
Chris
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Re: [Asterisk-Users] Polycom IP600 Phantom Ringing

[Jerry]

Never had any of my 100 or so act like that. What version of code are 
you running? I think 1.4.1 is the latest.

On Mar 8, 2005, at 8:05 PM, Ben Ruset wrote:
Hello list:
I have a very odd problem. Seemingly randomly, my Polycom IP600 phones 
will ring without a call being placed to it.

That is to say, a random phone will ring. Nothing shows up under 
Caller ID. Even the buttons that light up to show an incoming call do 
not light up. If you pick up the handset, you can hear the phone ring 
through the speaker.

Hanging up the phone makes it stop ringing. Then, sometime later, it 
will happen on another random extension.

Is this a common problem? Where can I look to start diagnosing this?
Thanks!
-ben
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RE: [Asterisk-Users] Dial() out and offer a menu system

[vgrskovic]

Hi Rauol,

Look at http://www.voip-info.org/wiki-Asterisk+auto-dial+out

Basically you need to create a .CALL file in
/var/spool/asterisk/outgoing

Set the context in the .CALL file to [mainmenu] and your set

-Vinko

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Raoul
Bönisch
Sent: Tuesday, March 08, 2005 9:13 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dial() out and offer a menu system

Hello all!

I'd like my * to call out to somebody and offer the called party
a menu system. Everything should just be as if the called party
had initiated the call themselves.

This is my try:

exten => 100,1,Dial(Modem/g1:0555321)
exten => 100,2,Goto(mainmenu,s,1)

This doesn't really work, because the Dial cmd blocks further
execution until the called party hangs up.

Next try:

[macro-incom]
exten => s,1,Goto(mainmenu,s,1)

exten => 100,1,Dial(Modem/g1:0555321,7,gtTHM(incom))

Doesn't work, too. The line is just hung up by * I suppose.

How would I be able to do this?

Greetings!

Raoul

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Re: [Asterisk-Users] NAT Far End Traversal

[Cameron Beattie]

I am very interested in this idea, mainly from the "protection from a 
hostile telco" point of view. What are the problems that this is likely to 
cause: additional overhead, delay, firewall problems etc?

Perhaps this is a little off-topic and should be discussed in another forum 
but if people are interested I would appreciate your input.

Cameron
---
Original message
Date: Wed, 09 Mar 2005 08:11:39 +0800
From: Leo Ann Boon <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] NAT Far End Traversal
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed



Another question... Are you aware of a SIP ATA or phone that has some

kind of VPN (i.e. PPTP) client embedded in? This would make the NAT

problem go away nicely and provide added security...
The Zulty's phones support VPN. Then again, many firewalls don't pass
through VPN traffic nicely. Would be cool if we can have a phone that
supports SSL VPNs like OpenVPN.
leo
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Re: [Asterisk-Users] Broadvoice-like company in Canada?

[Sergey Kuznetsov]

Their international rates are very high. We are planning to provide 
North American plan in very short time, but our rates
for SIP connections are wy lower. You can check it on our web-site: 
http://www.hitcalls.com
If you want to be connected via SIP, just drop me a few lines to my 
email. We can even configure IAX2 connection for your
* server, and provide a local 416 area code number. This is still in 
beta stage, but works very well. We have a beta-testers
for this service, and some of them even posting in this mailing list.

PS: Probably this is wrong list for some ads ( it's suppose to be in 
asterisk-biz), but if people looking for good rates, this is our 
pleasure to help.

--
All the Best!
Sergey.
=
Sergey Kuznetsov
President/CEO
High Intellectual Technologies, Inc.
  Web: http://www.hitcalls.com
   E-mail: [EMAIL PROTECTED]
Business phone: (416) 548-9700 ext 37

JR wrote:
Mobitus seems very cool.
Here is another question for the group:
Is it easy to use Vonage with an * box? Could I order service from 
them and not use the equip. they send?

-JR
On 8-Mar-05, at 9:34 PM, jurgen wrote:
Hi Justin,
I used to work with the fine people at Mobitus. (www.mobitus.com).
Give them a try. Last I looked, they have some kind of free trial
offer.
...jurgen
On Tue, 8 Mar 2005 20:38:13 -0500, JR <[EMAIL PROTECTED]> wrote:
Hey folks,
I am looking for a no frills bring-your-own-SIP device VOIP company
similar to Broadvoice.
Does anyone have any experience with VOIP in Canada?
-JR
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--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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Re: [Asterisk-Users] Broadvoice-like company in Canada?

[JR]

Mobitus seems very cool.
Here is another question for the group:
Is it easy to use Vonage with an * box? Could I order service from them 
and not use the equip. they send?

-JR
On 8-Mar-05, at 9:34 PM, jurgen wrote:
Hi Justin,
I used to work with the fine people at Mobitus. (www.mobitus.com).
Give them a try. Last I looked, they have some kind of free trial
offer.
...jurgen
On Tue, 8 Mar 2005 20:38:13 -0500, JR <[EMAIL PROTECTED]> wrote:
Hey folks,
I am looking for a no frills bring-your-own-SIP device VOIP company
similar to Broadvoice.
Does anyone have any experience with VOIP in Canada?
-JR
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--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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[Asterisk-Users] New Help Site - cut down on Mailing List questions

[Mike Sander]

This is a re-post as it was pointed out that I replied to a different
thread instead of creating a new post. Sorry for the additional traffic.
Mike

Dear All,

I understand the excitement surrounding a service like Asterisk, and how
easy it is to jump in and ask a heap of questions. I also know how
frustrating it can be dealing with a 200+ post per day mailing list as one
of the question "answerers".

When I discovered Asterisk, I had a lot of study to do, because there are
no "real-world" examples out there, just the trivial ones on the tiki and
in the manual.

I hope to propose a solution.

I have (in a small time) downloaded and set up a repositor where we should
all post our conf files, in an effort to get a big resource of a lot of
different setups that we know just "work". The program is simple, and
looks like crap and is a testiment to my programming skills (or lack
thereof). If anyone feels like re-coding or hosting this, let me know.

You can find this at:
asterconf.hopto.org (i think this has popups for the free DNS)
or home.exetel.com.au/azyc/asterconf

In the same philosophy as the GPL and wiki, it is open to all to search,
view and download the conf code, however to post and add new categories,
you must register. The site will not send you any mail or spam or
anything.

Of course, you should scrub your conf files for IP addresses and
user/secrets, but otherwise, please post as much as you like. Please also
include a description of the purpose of the post, and what type of service
it runs on, for better searching.


As a registered user, you are also free to add comments to other people's
code snippets (but not change the code), and add more categories and
sub-categories. I have started by creating categories for the most common
conf files, under both working and broken sections.

As a new (or old) asterisk user, if you are stuck, feel free to post your
conf in the broken section and hopefully someone will come to help you.
This will stop people posting codes to this list and flodding it.


If we all use this resource, the we will reduce the amount of posts for
people looking for "instant" setups, who don't want to use AMP or
otherwise.

That way, we can return this list to the discussion of Asterisk issues,
rather than just a startup resource and helpdesk.


I'm always interested in anyone's comments.

Cheers

Mike Sander
sanderm at iprimus.com.au
+61 2 401 010 289 (Australian mobile)


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Re: [Asterisk-Users] All Circuits are Busy Now

[Pulu 'Anau]

One thing you have to do is take the 9 out of the extension before you send it
on to broadvoice.  exten:1 on your dial cmd there.

Pulu




Afe.to ANTS
POB 1478
Nuku'alofa, Tonga
Ph: Country code 676 - 27946 or 878-1332
http://www.afe.to
http://svcs.affero.net/rm.php?r=pulu


Quoting Scott Wolfe <[EMAIL PROTECTED]>:

> I have downloaded and installed [EMAIL PROTECTED] and I have installed X-Lite 
> on
> my Windows machine and I am able to connect it to the Asterisk server. I went
> ahead an created an account on Broadvoice today and followed the directions
> on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and
> http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when
> ever I try and make a call from Xlite I get the all circuits are Busy now
> recording.
>
> Do I need to create a Trunk or get rid of the one that's there? Currently
> listed is  the
> ZAP/g0 wich I think is for a hard line. Here is my current sip.conf and
> extensions.conf
>
> Thanks for any tips.
>   -Scott
>
>
>
> == sip.conf  ==
>
> ; Note: If your SIP devices are behind a NAT and your Asterisk
> ;  server isn't, try adding "nat=1" to each peer definition to
> ;  solve translation problems.
>
> [general]
>
> port = 5060   ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
> disallow=all
> allow=ulaw
> allow=alaw
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
>
> #include sip_nat.conf
> #include sip_additional.conf
>
> register =>
> [EMAIL PROTECTED]:pp:[EMAIL PROTECTED]/2197
>
> [sip.broadvoice.com]
> type=peer
> user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=xx
> secret=pp
> username=xx
> insecure=very
> context=from-broadvoice
> authname=xx
> dtmfmode=inband
> dtmf=inband
> authuser=xx
> ;Disable canreinvite if you are behind a NAT
> canreinvite=no
> quality=yes
>
> === Extensions.conf ===
> ; I only addedd:
>
> [VOIP-OUT]
> exten => _9NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30)
> exten => _9NXXNXX, 2, congestion()
> exten => _9NXXNXX, 102, busy()
>
>


-
Webmail provided by AFE.TO Ants
http://www.afe.to

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[Asterisk-Users] Could dialing long extensions be a problem?

[Ronald Wiplinger]

I have two DIDs from NuFone.
I have for the first DID a short extension (601 = 3 digits) and on the 
other one a long extension (886212345678 =12 digits)

Extensions.conf looks like:
[fromNuFone]
exten => 888xxx,1,Dial(SIP/601,20,tr)
exten => 888xxx,2,Dial(SIP/602,20,tr)
exten => 888xxx,3,hangup
;
;
exten => 866xxx,1,Dial(SIP/886212345678,60,tr)
exten => 866xxx,2,Voicemail,u886212345678   
exten => 866xxx,103,Voicemail,b886212345678
exten => 866xxx,104,hangup

first one works, second one not
*CLI shows:
   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-18", 
"SIP/601|20|tr") in new stack
   -- Called 601
   -- SIP/601-de44 is ringing
...
   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-10", 
"SIP/886212345678|60|tr") in new stack
Mar  9 09:31:38 WARNING[29209]: chan_sip.c:1405 create_addr: No such 
host: 886212345678
Mar  9 09:31:38 NOTICE[29209]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3)

changing to a short extension works!

However, dialing from 601 to 886212345678
[long_extensions]
exten => 886212345678,1,Dial(SIP/886212345678,60,tr)
exten => 886212345678,2,NoOp(${DIALSTATUS})
exten => 886212345678,3,Voicemail,u886212345678  
exten => 886212345678,104,Voicemail,b886212345678
exten => 886212345678,105,hangup

  -- Executing Dial("SIP/615-610d", "SIP/886212345678|60|tr") in new stack
   -- Called 886212345678
   -- SIP/886212345678-e925 is ringing
How can I solve it?
bye
Ronald


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[Asterisk-Users] All Circuits are Busy Now

[Scott Wolfe]

I have downloaded and installed [EMAIL PROTECTED] and I have installed X-Lite 
on my Windows machine and I am able to connect it to the Asterisk server. I went 
ahead an created an account on Broadvoice today and followed the directions on 
http://voip-info.org/wiki-Asterisk+settings+Broadvoice and 
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but 
when ever I try and make a call from Xlite I get the all circuits are Busy now 
recording. 
 
Do I need to create a Trunk or get rid of the one 
that's there? Currently listed is  the  
ZAP/g0 wich I think is for a hard line. Here is my 
current sip.conf and extensions.conf
 
Thanks for any tips. 
  -Scott
 
 
 
== sip.conf  
==
 
; Note: If your SIP devices are behind a NAT and 
your Asterisk;  server isn't, try adding "nat=1" to each peer 
definition to;  solve translation problems.
 
[general]
 
port = 
5060   ; Port to bind 
to (SIP is 5060)bindaddr = 0.0.0.0    ; Address to bind to 
(all addresses on 
machine)disallow=allallow=ulawallow=alawcontext = 
from-sip-external ; Send unknown SIP callers to this contextcallerid = 
Unknown
 
#include sip_nat.conf#include 
sip_additional.conf
 
register => xx@sip.broadvoice.com:pp:[EMAIL PROTECTED]/2197
 
[sip.broadvoice.com]type=peeruser=phonehost=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=xxsecret=ppusername=xxinsecure=verycontext=from-broadvoiceauthname=xxdtmfmode=inbanddtmf=inbandauthuser=xx;Disable 
canreinvite if you are behind a 
NATcanreinvite=noquality=yes
=== Extensions.conf ===
; I only addedd:
 
[VOIP-OUT]exten => _9NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) 
exten => _9NXXNXX, 2, congestion() exten => _9NXXNXX, 102, 
busy()
 
 
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[Asterisk-Users] Dial() out and offer a menu system

[Raoul Bönisch]

Hello all!

I'd like my * to call out to somebody and offer the called party
a menu system. Everything should just be as if the called party
had initiated the call themselves.

This is my try:

exten => 100,1,Dial(Modem/g1:0555321)
exten => 100,2,Goto(mainmenu,s,1)

This doesn't really work, because the Dial cmd blocks further
execution until the called party hangs up.

Next try:

[macro-incom]
exten => s,1,Goto(mainmenu,s,1)

exten => 100,1,Dial(Modem/g1:0555321,7,gtTHM(incom))

Doesn't work, too. The line is just hung up by * I suppose.

How would I be able to do this?

Greetings!

Raoul

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[Asterisk-Users] Polycom IP600 Phantom Ringing

[Ben Ruset]

Hello list:
I have a very odd problem. Seemingly randomly, my Polycom IP600 phones 
will ring without a call being placed to it.

That is to say, a random phone will ring. Nothing shows up under Caller 
ID. Even the buttons that light up to show an incoming call do not light 
up. If you pick up the handset, you can hear the phone ring through the 
speaker.

Hanging up the phone makes it stop ringing. Then, sometime later, it 
will happen on another random extension.

Is this a common problem? Where can I look to start diagnosing this?
Thanks!
-ben
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[Asterisk-Users] Broadvoice-like company in Canada?

[JR]

Hey folks,
I am looking for a no frills bring-your-own-SIP device VOIP company 
similar to Broadvoice.

Does anyone have any experience with VOIP in Canada?
-JR
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[Asterisk-Users] Sip 400 bad request - broadvoice error

[Zanzamar Majere]

I have searched the list and cannot find a sip 400 solution posted that
solves my problem.  If anyone has any thoughts or suggestions on the
following I would greatly appreciate it.

I didn't have this error before Broadvoice made their changes this
weekend.  Now when I make a call it connects but, I cannot hear anything
on the other end... 


The full message I have is:

8 headers, 0 lines
Sending to 147.135.4.128 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
147.135.4.128:5060;branch=z9hG4bK2ib80u200o5079o6q7c1.1sr
From:
;tag=SD50bgc99-1538429642-1110330497384
To: "Waiting Room/Caribou Insurance"
;tag=as49ead6ef
Call-ID: [EMAIL PROTECTED]
CSeq: 40722885 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 147.135.4.128:5060
Destroying call '[EMAIL PROTECTED]'
server*CLI>

Sip read:
SIP/2.0 400 Bad request
Via: SIP/2.0/UDP 67.42.244.202:5060;branch=z9hG4bK2d71d419
From: "asterisk" ;tag=as49ead6ef
To:
;tag=SD50bgc99-1929245493-1110330503603
Call-ID: [EMAIL PROTECTED]
CSeq: 107 INVITE
Content-Length: 0

My SIP.conf is:

register =>
@sip.broadvoice.com::@sip.broadvoice.com


[zachphone]
type=friend
[EMAIL PROTECTED]
password=.asd.fgh.
host=dynamic
dtmfmode=inband
defaultip=192.168.55.65
mailbox=12
callerid="Zach/Caribou Insurance" <303.557.0057x12>
context=sip ;your context in extensions.conf

[sip.broadvoice.com]
type=friend
username=3035570057
fromuser=__
authname=__
fromdomain=sip.broadvoice.com  (sip.broadvoice.com is configured in my
/etc/hosts to point to proxy.lax.broadvoice.com's ip)
secret=--
host=sip.broadvoice.com
port=5060
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=glaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no

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Re: [Asterisk-Users] Question about AGI vs. FastAGI vs. straight C/DB development

[Mohit Muthanna]

> Opinions? Would any of you bother writing an IVR-only
> application (like the app_voicemail application) in C?
> Why or why not?

I would strongly advise against using C _unless absolutely necessary_.
While it is true that you will see performance gains by using C
(linked in that is), the tradeoff from using a scripting language
(like Perl or Python) is quite significant.

What tradeoff? Maintenability, development effort / ease, portability, security.

Of course, it can be argued that all of the above can be overcome by a
good C programmer. But that takes a good, experienced C programmer;
and even so, you may be better off using a scripting language. C
definitely has it's place, but this, IMHO, is not the right one.
Also... debugging memory leaks, maintaining build files / makefiles,
etc. is just not fun.

If performance is your primary concern, I would suggest using FastAGI.
An added benefit to FastAGI is the ability to easily write load
balanced and fault tolerant apps. This makes your applications easier
to scale up as the need arises.

The problem with just AGI (not FastAGI), which really is more relevant
to systems with lots of short concurrent calls, is that the OS has to
fork a new process for every call. For some applications, this can
have a significant performance impact.

If your FastAGI app is implemented correctly, e.g., by passing
connections to new threads, or using a poll()/select() style loop, you
can see considerable speed improvements.

Anyhow... my point is... don't use C unless you have a good reason. Go
for Perl / Python w/ FastAGI.

My $0.02.

HTH,
Mohit.



-- 
Mohit Muthanna [mohit (at) muthanna (uhuh) com]
"There are 10 types of people. Those who understand binary, and those
who don't."
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Re: [Asterisk-Users] where is voice conduits

[Tim Mattison]

Well, they took my money, I posted that they were a scam, a bunch of
people told me to stop being alarmist and posting crap, and they still
have my money.

As far as service goes... yeah, there's none of that.

On Mon, 2005-02-28 at 21:32 -0500, ross jones wrote:
> on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote:
> 
> > 
> > There was a thread a month or two ago on here about voiceconduits. The
> > general gist was they are not yet open for public business.
> 
> Are there any voice conduits customers out there?  if not, maybe I ought to
> just walk away.  
> 

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RE: [Asterisk-Users] STOP NOW not responding

[Marios Andreou]

I have the same experience and it seems mpg123 is hanging and asterisk is 
waiting on it.
Try killing mpg123 and see if asterisk will continue. (Actually exit)

Even if I do a restart now the same thing happens and I have to kill mpg123 and 
* restarts normally ???

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Klein
Sent: Tuesday, March 08, 2005 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] STOP NOW not responding

Try ps -auxww, find the process and kill it. Or, if you're on 
a system that supports killall, just killall asterisk.

safe_asterisk should restart the * process automatically, or if you're not 
running safe_asterisk, then just start the process after you've killed 
it.. I have no idea why STOP NOW doesn't work every time, but I 
experienced the problem with a 2004 CVS-HEAD update.

-Matt

-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us again."
- Hunter S. Thompson on the 2004 election.

On Tue, 8 Mar 2005, Wiley Siler wrote:

> Has anyone had any new information about STOP NOW hanging?  I am using
> [EMAIL PROTECTED] 0.6 and today my system just stopped responding.  I issued
> the usual STOP NOW command and it just returns to the CLI.  I have found
> a lot of info regarding others having this happen but nothing that
> addresses resolution.  I do not do a lot of calls so I am not sure why
> this would occur.  Any info woud be appreciated.  I am continuing to
> search the Wiki and list so if I find something I will post it.
>
> Thanks,
> Wiley
>
>
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Re: [Asterisk-Users] STOP NOW not responding

[Matt Klein]

Try ps -auxww, find the process and kill it. Or, if you're on 
a system that supports killall, just killall asterisk.

safe_asterisk should restart the * process automatically, or if you're not 
running safe_asterisk, then just start the process after you've killed 
it.. I have no idea why STOP NOW doesn't work every time, but I 
experienced the problem with a 2004 CVS-HEAD update.

-Matt
-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us again."
- Hunter S. Thompson on the 2004 election.
On Tue, 8 Mar 2005, Wiley Siler wrote:
Has anyone had any new information about STOP NOW hanging?  I am using
[EMAIL PROTECTED] 0.6 and today my system just stopped responding.  I issued
the usual STOP NOW command and it just returns to the CLI.  I have found
a lot of info regarding others having this happen but nothing that
addresses resolution.  I do not do a lot of calls so I am not sure why
this would occur.  Any info woud be appreciated.  I am continuing to
search the Wiki and list so if I find something I will post it.
Thanks,
Wiley

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Re: [Asterisk-Users] NAT Far End Traversal

[Leo Ann Boon]

Another question... Are you aware of a SIP ATA or phone that has some 
kind of VPN (i.e. PPTP) client embedded in? This would make the NAT 
problem go away nicely and provide added security...
The Zulty's phones support VPN. Then again, many firewalls don't pass 
through VPN traffic nicely. Would be cool if we can have a phone that 
supports SSL VPNs like OpenVPN.

leo
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Re: [Asterisk-Users] Asterisk Interop w/ Level 3

[William Suffill]

Seems to be a popular move on this list I'm sure some of those that
have taken the plunge already could be of assistance.

LiveVoip/Teliax/Netlogic are 3 that I've heard use L3 currently that
are on this list. Probably more of a -biz question though then the
general user population.

-- William
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Re: [Asterisk-Users] Nortel ATA not passing dtmf tones to fxo

[Carlos Chavez]

> Yes the norstar is a piece of work. We have the same problem. The 
> only solution is to dial asterisk then press feature 808 on your 
> digital phone then the phone system will pass the dtmf - what a crock!
> 

 How do you press feature 808 on the phone?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] STOP NOW not responding

[Wiley Siler]

Title: STOP NOW not responding






Has anyone had any new information about STOP NOW hanging?  I am using [EMAIL PROTECTED] 0.6 and today my system just stopped responding.  I issued the usual STOP NOW command and it just returns to the CLI.  I have found a lot of info regarding others having this happen but nothing that addresses resolution.  I do not do a lot of calls so I am not sure why this would occur.  Any info woud be appreciated.  I am continuing to search the Wiki and list so if I find something I will post it.

Thanks,

Wiley



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[Asterisk-Users] Voicetronix Tones

[Rod Bacon]

I'm sorry if this has been answered before. I seem to remember reading a 
similar thread a while back, but for the life of me I can't find it 
(after 30 minutes of intensive googling).

I have a voicetronix openline 4 card in an * server running CVS HEAD as 
of 1st March. Everything is working pretty much as as expected (with a 
little more echo than zaptel?).

The question I have is regarding tones heard when dialling. When a sip 
phone dials a PSTN number, a dialtone is heard after the number is 
dialled, before the ringing tone. I assume that this dialtone is coming 
from the voicetronix card.

Can this additional dialtone be bypassed? It's rather annoying.
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Re: [Asterisk-Users] budgetphone

[Michiel van Baak]

On 00:09, Wed 09 Mar 05, Jasper Spaans wrote:
> FYI: I just set up
> http://www.voip-info.org/tiki-index.php?page=Talkin2ya which describes
> how to setup Asterisk for usage with talkin2ya.
> 
> Works for Michiel & me.
> 
> Cheers,
> 
> Jasper
> 

Works like a charm.
Just called my mom and she didn't even notice.
And as you might know moms are always asking about your food
program when you sound different ;)

Thnx Jasper
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

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Re: [Asterisk-Users] budgetphone

[Jasper Spaans]

FYI: I just set up
http://www.voip-info.org/tiki-index.php?page=Talkin2ya which describes
how to setup Asterisk for usage with talkin2ya.

Works for Michiel & me.

Cheers,

Jasper

On Sat, 5 Mar 2005 00:12:55 +0100, Roman Zhovtulya
<[EMAIL PROTECTED]> wrote:
> Yes, I've had this before with sipgate.
> 
> Try using either "31557110304" or "557110304" in both places in:
> 
> register => 31557110304:[EMAIL PROTECTED]/557110304
> 
> And use use this number as a context for incoming calls
> 
> What also might work: the incoming number (557110304) in
> register => 31557110304:[EMAIL PROTECTED]/557110304
> 
> Should be the same as a context name:
> 
> [31557110304]
> type=friend
> context=from-budgetphone
> host=sip.budgetphone.nl
> username=31557110304
> secret=my_budgetphone_pass
> qualify=yes
> nat=yes
> canreinvite=no
> insecure=very
> 
> Hope it helps.
> 
> Regards,
> Roman Zhovtulya
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Michiel
> van Baak
> Sent: Freitag, 4. März 2005 18:53
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] budgetphone
> 
> Hi all,
> 
> I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving
> calls works like a charm, I even redirected my normal PSTN number to the
> number I got from them so everything ends up in my * server. Before I
> ask them to take over my normal phone number I wanted to test all of it,
> so I ordered some calling minutes to test. Now I cannot get outbound
> calling to work with them. Anyone here knows how to set it up ?
> 
> Some more info:
> Asterisk CVS-HEAD as of 15-02-2005
> 
> My sip.conf
> 
> [general]
> context=from-sip
> realm=vanbaak
> port=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> maxexpirey=3600
> defaultexpirey=120
> musicclass=default
> allow=all
> language=en
> relaxdtmf=yes
> rtptimeout=60
> rtpholdtimeout=300
> ;trustrpid = no
> ;progressinband=no
> useragent=Asterisk
> nat=no
> externip=XXX.XXX.XXX.XXX
> localnet=192.168.2.0/255.255.255.0
> promiscredir = no
> register => 7304502:[EMAIL PROTECTED]/7304502
> register => 31557110304:[EMAIL PROTECTED]/557110304
> register => mvanbaak:[EMAIL PROTECTED]
> 
> [7304502]
> type=friend
> context=from-sipgate
> host=sipgate.de
> username=7304502
> secret=my_sipgate_pass
> nat=yes
> canreinvite=no
> insecure=very
> 
> [31557110304]
> type=friend
> context=from-budgetphone
> host=sip.budgetphone.nl
> username=31557110304
> secret=my_budgetphone_pass
> qualify=yes
> nat=yes
> canreinvite=no
> insecure=very
> 
> [nikotel]
> secret=my_nikotel_pass
> username=mvanbaak
> fromuser=mvanbaak
> type=peer
> context=from-nikotel
> host=calamar0.nikotel.com
> canreinvite=no
> nat=yes
> 
> ...some more entries for sip phones/softphones follow, they
> all work...
> 
> the dial statement in my extensions.conf
> 
> [outgoing-budgetphone]
> exten => _0X,1,SetAccount(outgoing-budgetphone)
> exten => _0X,2,SetCallerID(31557110304)
> exten => _0X,3,Dial(SIP/31557110304/${EXTEN})
> exten => _0X,4,Congestion
> exten => _0X,104,Busy
> 
> And this is wat I get on the CLI when I call my cellphone:
> 
> -- Executing SetAccount("SIP/michiel-d5bd", "outgoing-budgetphone")
> in new stack
> -- Executing SetCallerID("SIP/michiel-d5bd", "31557110304") in new
> stack
> -- Executing Dial("SIP/michiel-d5bd", "SIP/31557110304/06X")
> in new stack
> -- Called 31557110304/06X
> Mar  4 18:51:11 WARNING[4529]: chan_sip.c:6830 handle_response:
> Forbidden - wrong password on authentication for INVITE to
> '"31557110304" ;tag=as0ccbacfe'
> -- SIP/31557110304-5857 is circuit-busy
>   == Everyone is busy/congested at this time
> -- Executing Busy("SIP/michiel-d5bd", "") in new stack
>   == Spawn extension (internal, 06X, 104) exited non-zero on
> 'SIP/michiel-d5bd'
> -- Got SIP response 483 "Too many hops" back from 81.23.228.150
> 
> I tripple checked my password, and I am sure it is correct.
> 
> What to do ?
> 
> --
> Michiel van Baak
> http://lunteren.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
> 
> "Two of the most famous products of Berkeley are LSD and BSD. I don't
> think that this is a coincidence."
> 
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-- 
Jasper Spaans   http://jsp.vs19.net/
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Aster

Re: [Asterisk-Users] Adit 600 for asterisk

[Jerry]

On Mar 8, 2005, at 2:14 PM, Steven Critchfield wrote:
On Tue, 2005-03-08 at 14:07 -0600, Dennis Webb wrote:
Ok,  I've pretty much decided to try the Adit route.  Somebody who has
experience with these tell me if I'm missing something.
I have 15 incoming PSTN lines.  T1 is not an option at current
location.  I want to put in an Adit 600 with 2 8-port FXO boards.  The
adit will then connect to * via a digium t1 board.  I configure
zaptel.conf for the T1.  What other parts would be needed?  How do the
PSTN lines connect to the Adit, standard rj11 jacks? It looks to be
about a $2500 investment and I need to know if there is anything
special I am missing.
The Adit has a 50 pin D connector that is like the old SCSI connectors.
To connect your phone lines, you either want to get a punch down block
that has a 50 pin connector on it, or possibly a device with RJ11 style
jacks and a 50 pin connector. Then you connect it with a 25 pair cable
with 50 pin connectors on either side.
Go to any reputable supplier near you and they should be able to help
you look at and find what you are comfortable with to use for
installation.
--  
Install an RJ21 and cable to your Adit. As Steve mentioned the Adit has  
2 50pin connectors. They are female. An RJ21 is a 66M150 Block mounted  
on a bracket and wired to 2 50 pin connectors mounted on the same  
bracket. These connectors may be either male or female, but generally  
female.
See  
http://www.siemon.com/e-catalogXML/ 
datasheet.aspPN=157C&FAM=S66PrewiredMSerie&P=5739 for an example.

Go to your local telecom distributor, probably who you are ordering  
your Adit from, and get a prewired block and 2 25pr cables to connect  
your adit. If you are not going to ever exceed 24 lines then you may  
only connect 1 25pr able.

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Re: [Asterisk-Users] DID in the U.S.

[Doug Millsaps]

At 04:15 PM 3/8/2005, you wrote:
Hello!
Have a look at the following page:
  http://www.tex-an-2000.com/plxr.html
"Block of 10.000 DID numbers: No charge"
Is there something comparable in the LA area?
Andreas
I believe it's only free if you pay for the other services listed on that page. 

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Re: [Asterisk-Users] Broadvoice users...

[Dalon Westergreen]

I do not believe that BV restricts the number of outgoing calls, but i
did hear that there agreement states charge you for more then 4
simultaneous calls.  I have also heard that they have not done this to
date.

--Dalon


On Wed, 09 Mar 2005 08:18:50 +1100, Rod Bacon
<[EMAIL PROTECTED]> wrote:
> Do broadvoice limit the number of concurrent calls that any given sip
> registrant can make? What about other similar providers?
> 
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Re: [Asterisk-Users] Line-in as MOH source

[Rod Bacon]

Did you ever work this out?
- Original Message - 
From: "Niksa Baldun" <[EMAIL PROTECTED]>
Sent: Monday, January 03, 2005 1:07 PM
Subject: [Asterisk-Users] Line-in as MOH source


Hello,
Most traditional PBX-es have the ability to use external audio source
(e.g. radio tuner) for music on hold. This is also useful because you
can let your users listen to radio by dialing some extension.
I wanted to achieve the same on asterisk, and chan_alsa seemed the
logical choice. I installed ALSA drivers, connected the radio to line-in
and added the folowing to extensions.conf:
exten => *55,1,Dial(Console/Line)
And indeed, now I could listen to radio by dialing *55. There are some
problems, however:
1. Asterisk treats this as a normal call, so only one user can listen at
a time. Is there a way to let several user listen simultaneously?
2. As this is a low priority call, it should be dropped when incoming
call is sent to a channel which dialed it. I guess SoftHangup could be
used, but I don't know how to determine which channel to hangup.
3. I have no idea how to use this as a source for music on hold.
Any help would be appreciated.
Niksa

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[Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbing thegk file

[Jerry Geis]

SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gk file 
and the software version CP7912XXX file

The gk file must be lower case..
jerry
-
Where is SEPDEFAULT.cnf ?
Caps?

/ I have 8 cisco 7912 phones. 5 are working just fine.
/>/ They boot grab the gk file from the tftp server and
/>/ off to the races.
/>/ 
/>/ The other three phones are a no-go. I can view the tftp log and
/>/ see they are asking for the file. The file is there but these 3 phones
/>/ do not grab it.
/>/ 
/>/ As you can see below the file is asked for and the file is present.
/>/ The other 5 phones work.
/>/ 
/>/ Any idea on what might be going on? Thanks,
/>/ 
/>/ Jerry
/>/ -
/>/ Mar  8 10:43:56 mail dhcpd: DHCPREQUEST for 192.168.255.250 from 
/>/ 00:12:00:45:90:ba via eth1
/>/ Mar  8 10:43:56 mail dhcpd: DHCPACK on 192.168.255.250 to 
/>/ 00:12:00:45:90:ba via eth1
/>/ Mar  8 15:43:56 mail in.tftpd[25944]: RRQ from 192.168.255.250 filename 
/>/ SEP0012004590BA.cnf
/>/ Mar  8 15:43:56 mail in.tftpd[25945]: RRQ from 192.168.255.250 filename 
/>/ SEPDEFAULT.cnf
/>/ Mar  8 15:43:56 mail in.tftpd[25946]: RRQ from 192.168.255.250 filename 
/>/ gk0012004590ba
/>/ Mar  8 15:44:30 mail in.tftpd[25984]: RRQ from 192.168.255.250 filename 
/>/ SEP0012004590BA.cnf
/>/ Mar  8 15:44:30 mail in.tftpd[25985]: RRQ from 192.168.255.250 filename 
/>/ SEPDEFAULT.cnf
/>/ Mar  8 15:44:30 mail in.tftpd[25986]: RRQ from 192.168.255.250 filename 
/>/ gk0012004590ba
/>/ 
/>/ # ls -l /tftpboot/
/>/ total 860
/>/ drwxr-xr-x  2 root root   4096 Mar  8 16:54 7912
/>/ -rw-r--r--  1 root root 327307 Jan 13 14:59 CP7912010200SIP040406A.sbin
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk001200348afd
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004590ba
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk00120045944e
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004596a5
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004990ba
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk001200682ca9
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk00120093554f
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk001200a64e4b
/>/ -rw-r--r--  1 root root590 Mar  2 15:18 gk001200a650b0
/>/ -rw-r--r--  1 root root 13 Jan 12 14:26 OS79XX.TXT
/>/ -rw-r--r--  1 root root 476182 Jan 12 14:11 P0S3-05-1-00.bin
/>/ -rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03D1C.cnf
/>/ -rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03FD7.cnf
/>/ -rwxr-xr-x  1 root root   2098 Jan 13 15:53 SIP00124356C136.cnf
/>/ -rwxr-xr-x  1 root root   6040 Jan 12 14:27 SIPDefault.cnf/

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Re: [Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbing the gk file

[Rich Adamson]

Where is SEPDEFAULT.cnf ?
Caps?



> I have 8 cisco 7912 phones. 5 are working just fine.
> They boot grab the gk file from the tftp server and
> off to the races.
> 
> The other three phones are a no-go. I can view the tftp log and
> see they are asking for the file. The file is there but these 3 phones
> do not grab it.
> 
> As you can see below the file is asked for and the file is present.
> The other 5 phones work.
> 
> Any idea on what might be going on? Thanks,
> 
> Jerry
> -
> Mar  8 10:43:56 mail dhcpd: DHCPREQUEST for 192.168.255.250 from 
> 00:12:00:45:90:ba via eth1
> Mar  8 10:43:56 mail dhcpd: DHCPACK on 192.168.255.250 to 
> 00:12:00:45:90:ba via eth1
> Mar  8 15:43:56 mail in.tftpd[25944]: RRQ from 192.168.255.250 filename 
> SEP0012004590BA.cnf
> Mar  8 15:43:56 mail in.tftpd[25945]: RRQ from 192.168.255.250 filename 
> SEPDEFAULT.cnf
> Mar  8 15:43:56 mail in.tftpd[25946]: RRQ from 192.168.255.250 filename 
> gk0012004590ba
> Mar  8 15:44:30 mail in.tftpd[25984]: RRQ from 192.168.255.250 filename 
> SEP0012004590BA.cnf
> Mar  8 15:44:30 mail in.tftpd[25985]: RRQ from 192.168.255.250 filename 
> SEPDEFAULT.cnf
> Mar  8 15:44:30 mail in.tftpd[25986]: RRQ from 192.168.255.250 filename 
> gk0012004590ba
> 
> # ls -l /tftpboot/
> total 860
> drwxr-xr-x  2 root root   4096 Mar  8 16:54 7912
> -rw-r--r--  1 root root 327307 Jan 13 14:59 CP7912010200SIP040406A.sbin
> -rw-r--r--  1 root root590 Mar  2 15:18 gk001200348afd
> -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004590ba
> -rw-r--r--  1 root root590 Mar  2 15:18 gk00120045944e
> -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004596a5
> -rw-r--r--  1 root root590 Mar  2 15:18 gk0012004990ba
> -rw-r--r--  1 root root590 Mar  2 15:18 gk001200682ca9
> -rw-r--r--  1 root root590 Mar  2 15:18 gk00120093554f
> -rw-r--r--  1 root root590 Mar  2 15:18 gk001200a64e4b
> -rw-r--r--  1 root root590 Mar  2 15:18 gk001200a650b0
> -rw-r--r--  1 root root 13 Jan 12 14:26 OS79XX.TXT
> -rw-r--r--  1 root root 476182 Jan 12 14:11 P0S3-05-1-00.bin
> -rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03D1C.cnf
> -rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03FD7.cnf
> -rwxr-xr-x  1 root root   2098 Jan 13 15:53 SIP00124356C136.cnf
> -rwxr-xr-x  1 root root   6040 Jan 12 14:27 SIPDefault.cnf


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Re: [Asterisk-Users] Please help with install * SOLVED

[Victoria Alexandru]

I'm not registered with wiki, but I can tell what was
the mod:

In rhconfig.h, in line 43 you'll find "&& &&". I'll
try to email Mandrake people to have certitude but for
now what I did was to remove one pair of &. I believe
this is a typo, unless is something missing between &&
&&. Thats why I say I need to signal this to mandrake
and have a confirmation.

Regards, V.

--- Don Pobanz <[EMAIL PROTECTED]> wrote:
> Victoria Alexandru wrote:
> > Thanks anyone, I found the problem in rhconfig.h.
> > After the fix I successfully compiled zaptel.
> > V.
> 
> I also am trying to compile Zaptel on Mandrake
> 10.2beta3. I have seen 
> the same errors you were. What did you change in
> rhconfig.h?
> 
> Don Pobanz
> 
> Adding any info to the wiki would be helpful.
>
http://www.voip-info.org/wiki-Asterisk+Linux+Mandrake
> 
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[Asterisk-Users] RE:DID in the U.S.

[Jason Kawakami]

-Original Message-

Hello!

There is something I really don't get: As I ordered a PRI ISDN line in
Germany 
with DID, I had not to pay anything for a "DID number block", now I'm trying

to get a PRI ISDN in the U.S. (CA) and SBC wants to charge more than 200 
USD/month for numbers. I mean, this has nothing to do with DID, where 
everything that comes after the "base number" will be transmitted to the PBX

anyway. Wasn't DID invented to get rid of number blocks?

--you are being charged for the reservation of the numbers being dedicated
to you.  Each of those numbers represents revenue to the LEC so they are
damn well gonna charge you for reserving them.  Usually, they are $0.15 per
number in a block and the block size varies.

Jason Kawakami
www.optellabs.com   
Salt Lake City, UT




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[Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbing the gk file

[Jerry Geis]

I have 8 cisco 7912 phones. 5 are working just fine.
They boot grab the gk file from the tftp server and
off to the races.
The other three phones are a no-go. I can view the tftp log and
see they are asking for the file. The file is there but these 3 phones
do not grab it.
As you can see below the file is asked for and the file is present.
The other 5 phones work.
Any idea on what might be going on? Thanks,
Jerry
-
Mar  8 10:43:56 mail dhcpd: DHCPREQUEST for 192.168.255.250 from 
00:12:00:45:90:ba via eth1
Mar  8 10:43:56 mail dhcpd: DHCPACK on 192.168.255.250 to 
00:12:00:45:90:ba via eth1
Mar  8 15:43:56 mail in.tftpd[25944]: RRQ from 192.168.255.250 filename 
SEP0012004590BA.cnf
Mar  8 15:43:56 mail in.tftpd[25945]: RRQ from 192.168.255.250 filename 
SEPDEFAULT.cnf
Mar  8 15:43:56 mail in.tftpd[25946]: RRQ from 192.168.255.250 filename 
gk0012004590ba
Mar  8 15:44:30 mail in.tftpd[25984]: RRQ from 192.168.255.250 filename 
SEP0012004590BA.cnf
Mar  8 15:44:30 mail in.tftpd[25985]: RRQ from 192.168.255.250 filename 
SEPDEFAULT.cnf
Mar  8 15:44:30 mail in.tftpd[25986]: RRQ from 192.168.255.250 filename 
gk0012004590ba

# ls -l /tftpboot/
total 860
drwxr-xr-x  2 root root   4096 Mar  8 16:54 7912
-rw-r--r--  1 root root 327307 Jan 13 14:59 CP7912010200SIP040406A.sbin
-rw-r--r--  1 root root590 Mar  2 15:18 gk001200348afd
-rw-r--r--  1 root root590 Mar  2 15:18 gk0012004590ba
-rw-r--r--  1 root root590 Mar  2 15:18 gk00120045944e
-rw-r--r--  1 root root590 Mar  2 15:18 gk0012004596a5
-rw-r--r--  1 root root590 Mar  2 15:18 gk0012004990ba
-rw-r--r--  1 root root590 Mar  2 15:18 gk001200682ca9
-rw-r--r--  1 root root590 Mar  2 15:18 gk00120093554f
-rw-r--r--  1 root root590 Mar  2 15:18 gk001200a64e4b
-rw-r--r--  1 root root590 Mar  2 15:18 gk001200a650b0
-rw-r--r--  1 root root 13 Jan 12 14:26 OS79XX.TXT
-rw-r--r--  1 root root 476182 Jan 12 14:11 P0S3-05-1-00.bin
-rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03D1C.cnf
-rwxr-xr-x  1 root root   3226 Jan 13 15:53 SIP000FF7C03FD7.cnf
-rwxr-xr-x  1 root root   2098 Jan 13 15:53 SIP00124356C136.cnf
-rwxr-xr-x  1 root root   6040 Jan 12 14:27 SIPDefault.cnf
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Re: [Asterisk-Users] Broadvoice users...

[Rich Adamson]

> Do broadvoice limit the number of concurrent calls that any given sip 
> registrant can make? What about other similar providers?

Each ITSP has a limit of some sort with their 'unlimited' plans
(usually one or two), and many of those with rates based on $/min 
typically allow some larger number of concurrent calls. Some
state what their limits are while others never mention it.

In the end, there's no free lunch.


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Re: [Asterisk-Users] RE: Re: MGCP to Inter Tel system

[Gary Carr]

-this is very true, however, the current version of the Axxess software
(9.0) supports SIP trunking natively on the IPRC.  I just got my Axxess
upgraded and am salivating to get * connected to it.

Hmm, so 9.0 is out and it supports SIP natively. How did you plan to 
integrate the 2?


Gary
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Re: [Asterisk-Users] The ZIP2 (was Recommended Phone for beginner)

[John Novack]

Mark Eissler wrote:
6) Zultsys ZIP 2 IP Phone ($94):
   - haven't seen much written about these either
   - they look pretty cheap
I bought one of these to use with Asterisk
Phone is fairly light, and the little rubber feet come off easily.
No display at all, so no back light issue
No speakerphone. On hook dialing though.
The handset is a funky shape
two line appearances, not with individual buttons, but toggle between 
calls and conference two calls does work as advertised.
SUPPOSED to support PoE, but have yet to try that.
Has an acoustic echo option
The web based setup seems fairly complete, and the user manual, 
available on line, is probably not worth printing, as all it does is 
commit to paper the web interface. It did explain what the AEC option 
meant, but little beyond making it worth reading or printing 100 pages.
Have yet to be able to get the MWI to work, but can't be sure that is 
the phone.
Support people are not too smart. I asked about a wall mount bracket for 
the phone, as it has a reversible clip that would allow the handset to 
hang on hook if it were wall mounted, and the base has notches to accept 
some sort of molded bracket, but support said " go to your telephone 
supply house"
Sure glad I didn't ask them a really hard question.

Would I buy another? Probably not. A buddy has one of the really cheap  
BT 101 phones, which seems to work really well. The speakerphone suffers 
from echo, but other than that, for light duty use it probably is a 
better choice.

Personally I don't think any SIP phone needs to cost much over 100 
bucks. and could still be usable and rugged.

JMO
John Novack
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Re: [Asterisk-Users] DID in the U.S.

[Andreas Roedl]

Hello!

Am Dienstag, 8. März 2005 22:38 schrieb Rich Adamson:
> The different US telephone companies have many different ideas on how
> to generate revenue, and how they do that varies dramatically between
> the different companies.
>
> At one time, the State of Iowa had 600 "independent" telephone companies
> and most likely hundreds of rates for the same thing/service. That
> has obviously decreased, but there is nothing limiting each telephone
> company from doing whatever they want for many different services.
> Even basic isdn prices vary dramatically across the US.
>
> Got to wonder how some ITSPs obain large blocks of numbers and
> essentially give them away, while others charge large amounts for
> the same thing. I'm sure they've done their research. ;)

Have a look at the following page:

  http://www.tex-an-2000.com/plxr.html

"Block of 10.000 DID numbers: No charge"

Is there something comparable in the LA area?


Andreas
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Re: [Asterisk-Users] Asterisk Interop w/ Level 3

[George Pajari]

Mr. Willingham wrote:
Has anyone done interop testing with Level 3 and Asterisk.  If so, 
would you be willing to share your experiences.
We're in the queue waiting for Level 3 to start the process. Would love 
to exchange email with those who have gone before or share the pain with 
others going through it now. Note that the process (at least for us) is 
covered by an NDA we have with Level 3 and so the conversations would 
have to be general in nature or cleared with Level 3. I will go back and 
re-read the NDA to see exactly what it covers.

--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
 www.netvoice.ca  www.ip-centrex.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
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Re: [Asterisk-Users] Please help with install * SOLVED

[Don Pobanz]

Victoria Alexandru wrote:
Thanks anyone, I found the problem in rhconfig.h.
After the fix I successfully compiled zaptel.
V.
I also am trying to compile Zaptel on Mandrake 10.2beta3. I have seen 
the same errors you were. What did you change in rhconfig.h?

Don Pobanz
Adding any info to the wiki would be helpful.
http://www.voip-info.org/wiki-Asterisk+Linux+Mandrake
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RE: [Asterisk-Users] Cisco 7940 Upgrade Failing

[Rich Adamson]

How could I possibly be da man when you're da man. You fixed it!


> You da man!  Thanks.
> 
> 
> Regards,
> 
> Juan Staalenburg
> 
> -Original Message-
> 
> > Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above.
> > Can't get it to upgrade on its own via TFTP.  Phones w SCCP image will
> > upgrade fine but I can't get these 2.0.3s to start the firmware upgrade.
> 
> On some older 79x0's, we've had to delete a bunch of the config statements
> within the *.cnf files in order to upgrade. Deleting comments and the 
> majority of statements won't have any impact in terms of upgrading, just
> make sure you keep a full copy of the files around so that once the
> upgrade is complete, you know what parameters were there to start with.
> 
> The above is based on upgrading several used 79x0's (with experience).
> Seems not all phones were created equal (eg, maybe differences in buffer
> sizes, nvram, or something).
> 
> I never did try to figure out why some worked and some did not without
> deleting that stuff. 


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Re: [Asterisk-Users] DID in the U.S.

[Rich Adamson]

> There is something I really don't get: As I ordered a PRI ISDN line in 
> Germany 
> with DID, I had not to pay anything for a "DID number block", now I'm trying 
> to get a PRI ISDN in the U.S. (CA) and SBC wants to charge more than 200 
> USD/month for numbers. I mean, this has nothing to do with DID, where 
> everything that comes after the "base number" will be transmitted to the PBX 
> anyway. Wasn't DID invented to get rid of number blocks?

The different US telephone companies have many different ideas on how
to generate revenue, and how they do that varies dramatically between
the different companies. 

At one time, the State of Iowa had 600 "independent" telephone companies
and most likely hundreds of rates for the same thing/service. That
has obviously decreased, but there is nothing limiting each telephone
company from doing whatever they want for many different services.
Even basic isdn prices vary dramatically across the US.

Got to wonder how some ITSPs obain large blocks of numbers and 
essentially give them away, while others charge large amounts for
the same thing. I'm sure they've done their research. ;)


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Re: [Asterisk-Users] Nortel ATA not passing dtmf tones to fxo

[Steve Clark]

Carlos Chavez wrote:
 I am trying to integrate a Nortel Norstar system with an Asterisk service
using a TDM04B card (4 fxo).  So far everyting works from Asterisk to Nortel.
 The problem is when someone dials from the Nortel PBX to the Asterisk server.
 Asterisk answers the call and provides a dialtone (with DISA) but appartently
the DTMF tones are not passed to asterisk and the call cannot proceed.
This only happens when calling from a digital phone on the Nortel.  If I
connect an analog phone to the PBX and dial from there the call can go
through.  Anyone has experience with this?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Yes the norstar is a piece of work. We have the same problem. The only solution 
is to dial asterisk then press feature 808 on your digital phone then the phone 
system will pass the dtmf - what a crock!

HTH,
Steve
--
"They that give up essential liberty to obtain temporary safety,
deserve neither liberty nor safety."  (Ben Franklin)
"The course of history shows that as a government grows, liberty
decreases."  (Thomas Jefferson)

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Re: [Asterisk-Users] Options for Attendant Console.

[Paul Zimm]

www.quadrasoftware.com
Kyle
Will McCown wrote:
We've been playing with Asterisk with an eye towards possibly
replacing or augmenting our existing PBX serving about over 600
phones (and needing to expand).  The one missing bit that I can't
find any mention of is an Attendant Console.  Are there any
good solutions out there?
I've considered that maybe one of the better softphones might
suffice, but the ones I've looked at so far not geared to the
need to handle and quickly dispatch large numbers of calls.

www.asternic.org
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Re: [Asterisk-Users] DID in the U.S.

[Andreas Roedl]

Hello!

Am Dienstag, 8. März 2005 22:03 schrieb Joe Greco:
> > There is something I really don't get: As I ordered a PRI ISDN line in
> > Germany with DID, I had not to pay anything for a "DID number block", now
> > I'm trying to get a PRI ISDN in the U.S. (CA) and SBC wants to charge
> > more than 200 USD/month for numbers. I mean, this has nothing to do with
> > DID, where everything that comes after the "base number" will be
> > transmitted to the PBX anyway. Wasn't DID invented to get rid of number
> > blocks?
> >
> > Please enlighten me. Thanks.
>
> You're burning up numbers that could be allocated to other customers.
> There's an incentive for LEC's to discourage this by charging you for
> the extra numbers.  The NANPA is getting tight on numbers, and at the
> point where we have to move to 11-digit or 12-digit dialing instead of
> 10, there will be an immense amount of agony.
>
> So usually you don't see providers just handing out blocks of numbers
> for free.
>
> Perhaps this isn't the case in Germany.

It seems to be a little different in Germany, but I'm not sure. It all depends 
on the base number (don't know the technical term). If the base number is 
short enough, you can do whatever you want with the remaining part. For 
example: 611035 is the base number of NI and I can add up to 5 DID digits, 
because the maximum length of a phone number in Germany is 11 digits. So I 
can have thousands DID numbers. The bigger the company, the shorter the base 
number, it seems. Another example is Reuters Berlin, that has 28885 as their 
base number and they can have ten times more DID numbers, because their 
number is one digit shorter than ours.

> DID wasn't invented to "get rid of number blocks".  DID was invented
> to *allow* number blocks, by passing off the last digits of the dialled
> number to a PBX.

I understand.

> Without that, a PBX (or attendant) has to answer the 
> line and ask you for the extension you want.  That situation has the
> virtue of only burning a single number, but is viewed as uglier.

And that's what I'm trying to prevent.


Andreas
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 8, Issue 63

[chris]

>   [EMAIL PROTECTED] wrote:

> > Christopher,
> >
> > Just as a total guess, check to be sure the PoE portion
> >of his cat5 cable are not either grounded, or touching
> > each other.  Also, be sure he's not connected to a PoE
> >capeable switch since cisco is reverse polarity of the
> > PoE standard.
> > Niles
Niles,

Thanks for the guess. We don't have any PoE equipment in out setup. On further
review it seems the phone is sending power down the network cable... Definitly
a hardware problem. Seems to have fried a port on a Linksys workgroup switch.
(the port shows a link with nothing connected) but at least it didn't fry our
main Dell switches.

> >
>
> Also, this sounds like it needs to go to Cisco tech
> support not asterisk support...
>
>
Robert,
"asterisk supprt" Are you kidding me? What's with all the list cops? I
wanted to see if anyone had seen anything like this. I wan't demanding that
someone diagnose and repair my problem...

Man, this list is decaying fast!

~C



This message was sent using IMP, the Internet Messaging Program.

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RE: [Asterisk-Users] Nortel ATA not passing dtmf tones to fxo

[Nathan C. Smith]

I fought with this and made the same discovery you did.  

I'm not happy with my solution, but I have SPA-3000s set up in a "hotline"
mode.  when a call comes into them from the Nortel ATA it is automatically
routed to an Aserisk Extension.  I hope somebody else chimes in with an ATA
setting to change or something.

-Nate

-Original Message-
From: Carlos Chavez [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 08, 2005 3:30 PM
To: Asterisk
Subject: [Asterisk-Users] Nortel ATA not passing dtmf tones to fxo



 I am trying to integrate a Nortel Norstar system with an Asterisk
service using a TDM04B card (4 fxo).  So far everyting works from Asterisk
to Nortel.  The problem is when someone dials from the Nortel PBX to the
Asterisk server.  Asterisk answers the call and provides a dialtone (with
DISA) but appartently the DTMF tones are not passed to asterisk and the call
cannot proceed.

This only happens when calling from a digital phone on the Nortel.  If I
connect an analog phone to the PBX and dial from there the call can go
through.  Anyone has experience with this?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Nortel ATA not passing dtmf tones to fxo

[Carlos Chavez]

 I am trying to integrate a Nortel Norstar system with an Asterisk service
using a TDM04B card (4 fxo).  So far everyting works from Asterisk to Nortel.
 The problem is when someone dials from the Nortel PBX to the Asterisk server.
 Asterisk answers the call and provides a dialtone (with DISA) but appartently
the DTMF tones are not passed to asterisk and the call cannot proceed.

This only happens when calling from a digital phone on the Nortel.  If I
connect an analog phone to the PBX and dial from there the call can go
through.  Anyone has experience with this?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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Re: [Asterisk-Users] Please help with install * SOLVED

[Victoria Alexandru]

Thanks anyone, I found the problem in rhconfig.h.
After the fix I successfully compiled zaptel.
V.

--- Ron Wellsted <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Have you built your kernel on that machine?
> 
> The errors suggest that while the kernel sources are
> installed, the
> kernel has not been built.
> 
> Check on the exact procedure for your distribution.
> 
> HTH
> 
> - --
> Ron Wellsted
> http://www.wellsted.org.uk
> [EMAIL PROTECTED]
> FWD:519961  Gossiptel:9309811
> N 52.567623, W 2.137621
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.6 (GNU/Linux)
> Comment: Using GnuPG with Thunderbird -
> http://enigmail.mozdev.org
> 
>
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>
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> =rETO
> -END PGP SIGNATURE-
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Re: [Asterisk-Users] Wildcard X100P or TDM400P?

[Steven Critchfield]

On Tue, 2005-03-08 at 14:27 -0600, Rich Adamson wrote:
> > On Tue, 2005-03-08 at 13:24 -0600, Rich Adamson wrote:

> > > If that's the case, then why has bug 2023 been lurking without any such
> > > comments for many many months?
> > 
> > If it is lurking for 10 months it is because no one cares to read it. I
> > promise you I don't look at any bug unless it is mentioned on a list as
> > is pertaining to code I am running. I have so little time now, I don't
> > go looking for needles in haystacks that I don't need.
> 
> The only reason for questioning the above is that Mark (and several others)
> have contributed various comments to bug 2022 and 2023, but at no time
> have any of them ever mentioned comments relative to wav vs other file 
> formats, and effectively have left the bug unresolved.

Go look at the code for format_wav.c. I have code in there and have
actually looked at it myself.

This is at line 64 of my code base.
#define GAIN 2  /* 2^GAIN is the multiple to increase the volume by */

And this around line 480 in the wav_write function.
if (fs->buf) {
tmpi = f->data;
/* Volume adjust here to accomodate */
for (x=0;xdatalen/2;x++) {
tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN));
if (tmpf > 32767.0)
tmpf = 32767.0;
if (tmpf < -32768.0)
tmpf = -32768.0;
tmp[x] = tmpf;
tmp[x] &= ~((1 << GAIN) - 1);

f is the frame to be written. f->data is the audio data to be written.
The for loop just uses pointer math to traverse each sample of the
buffer and increase the volume of the sample with clipping control.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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[Asterisk-Users] At my wits' end: DTMF works locally, but ignored for incoming calls from IP telcos

[Roman Zhovtulya]

Hello,
Has anyone seen this before: Incoming call routing (based on DTMF,
entering an extension of a party to call) works locally, but fails when
the call is coming into the same context from the IP telco or from the
landline.

I got a local phone number from sipgate.de (Germany) and would like to
route my incoming calls to the interal user extensions based on what the
caller enters. 

The problems reported on Asterisk console: 
1) When calling through an IP telco using SJPhone softphone: 
"RFC9989 support incomplete. Turn off on client if possible".

2) When calling from a normal landline phone, the DTMF I type is simply
ignored and the extension goes to timeout, without giving any error
messages, as if I didn't enter anything at all


I've tried changing the DTMF mode for incoming context, etc, played with
all kinds of settings for several days already, but nothing helps.


Here is the context where the incoming calls are placed

[fhostaffmenu]
; this code is working locally, but not for the incoming calls from the
outside;

include => fhostaff
exten => s,1,Ringing
exten => s,n,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10
seconds
exten => s,n,BackGround(welcome) ; Play a congratulatory message
exten => s,n,WaitExten  ; Wait for an extension to be dialed.
exten => t,1,BackGround(timeout); Play a timeout message
exten => t,n,Hangup


Any suggestions are highly appreciated.

Thanks a lot,
Roman Zhovtulya

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[Asterisk-Users] Incoming Fax Service question

[Justin Newman]

If you need to dial additional digits after pickup, use the D(...) command
with Dial. Why not just send the call to another extension or DID?
To detect fax on the line, you can use NVFaxDetect or NVBackgroundDetect.
More information on the Tikiwiki.

http://www.voip-info.org/tiki-index.php?page=NVFaxDetect
http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect

Justin

-- 
Justin J. Newman
Newman Telecom, Inc.

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[Asterisk-Users] Broadvoice users...

[Rod Bacon]

Do broadvoice limit the number of concurrent calls that any given sip 
registrant can make? What about other similar providers?

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[Asterisk-Users] Asterisk Interop w/ Level 3

[Gene Willingham]

 

Has anyone done interop testing with Level 3 and Asterisk. 
If so, would you be willing to share your experiences.

 

Gene






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RE: [Asterisk-Users] Cisco 7940 Upgrade Failing

[Staalenburg, Juan]

You da man!  Thanks.


Regards,

Juan Staalenburg
Teksavers, Inc.
(512) 255-8395 x1002
AIM: juanteksavers


-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 08, 2005 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7940 Upgrade Failing



> Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above.
> Can't get it to upgrade on its own via TFTP.  Phones w SCCP image will
> upgrade fine but I can't get these 2.0.3s to start the firmware upgrade.

On some older 79x0's, we've had to delete a bunch of the config statements
within the *.cnf files in order to upgrade. Deleting comments and the 
majority of statements won't have any impact in terms of upgrading, just
make sure you keep a full copy of the files around so that once the
upgrade is complete, you know what parameters were there to start with.

The above is based on upgrading several used 79x0's (with experience).
Seems not all phones were created equal (eg, maybe differences in buffer
sizes, nvram, or something).

I never did try to figure out why some worked and some did not without
deleting that stuff. 


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Re: [Asterisk-Users] Please help with install *

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Have you built your kernel on that machine?
The errors suggest that while the kernel sources are installed, the
kernel has not been built.
Check on the exact procedure for your distribution.
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961  Gossiptel:9309811
N 52.567623, W 2.137621
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.6 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iQEVAwUBQi4T10tP/KMNOfRbAQJajAgAso8fLd3qYmBhgfzUBrMDQ8jDE/kWH/4r
jcTiVHcsMxbm1kBxAL5zF9X6rDVpUrISw3Z2vgC+saCVuMjk8FhMjWLIcbPIAJr5
UhNI2i+JeBoXYl4Zxr/Chyny8EbbcG5U4H/oRh8YCQNjpQFeg+rgCUpqehpVEBsL
zMuSxp5QOEOFhc4uuplABYmKdaxza26CDhTfETI55oHKBJjsdj84ZN/EhEcV0ZiP
CyzdgYgMqBR441d/nj2QENAYXWhTEKRp+rNjS4519+O88ZWUhlQp/9Rbqd9rUaRI
D8TNPTBNwrjVYHgBveJzGFI1sbBfF1vNBaXP+6FtUniBGtngMoqhiQ==
=rETO
-END PGP SIGNATURE-
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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Rich Adamson]

Since he can "call into the box", he is registering with BV. Otherwise he
would not be able to call in. So, his outgoing calls are messed up one
way or another. How about doing a "sip debug" while placing a call via
BV and post the results?


> Can you call anywhere or is this problem just with broadvoice? Is there
> any type of firewall like a netscreen or iptables configured in your
> setup, which may be blocking outbound UDP? Do you have a packet capture
> of the traffic that is leaving your network for broadvoice? You should
> try to inspect the signaling exchange with their proxy.
> 
> -Original Message-
> From: Jerry Geis [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, March 08, 2005 2:53 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Broadvoice latest changes and still not
> working
> 
> I tried removing the permit and that made no difference.
> 
> I can still call in to the box but no calls out.
> 
> Jerry
> 
> --
> 
> Yes it is working just fine for me with the same sip.conf that you have.
> ??
> Except the "permit=sip.broadvoice.com"
> You can see my config at
> http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
> 
> Also what is your extensions.conf ?
> 
> -Original Message-
> From: asterisk-users-bounces at lists.digium.com
> 
> [mailto:asterisk-users-bounces at lists.digium.com
> ] On Behalf Of
> James Taylor
> Sent: Tuesday, March 08, 2005 2:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
> working
> 
> Does anybody have Broadvoice outbound working?
> 
> On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis  >  
> wrote:
> 
> >/ Here is my configs. from a previous post...
> />/
> />/ Jerry
> />/
> />/ --
> />
> 
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---End of Original Message-


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Re: [Asterisk-Users] DID in the U.S.

[Joe Greco]

> Hello!
> 
> There is something I really don't get: As I ordered a PRI ISDN line in 
> Germany 
> with DID, I had not to pay anything for a "DID number block", now I'm trying 
> to get a PRI ISDN in the U.S. (CA) and SBC wants to charge more than 200 
> USD/month for numbers. I mean, this has nothing to do with DID, where 
> everything that comes after the "base number" will be transmitted to the PBX 
> anyway. Wasn't DID invented to get rid of number blocks?
> 
> Please enlighten me. Thanks.

You're burning up numbers that could be allocated to other customers.
There's an incentive for LEC's to discourage this by charging you for
the extra numbers.  The NANPA is getting tight on numbers, and at the
point where we have to move to 11-digit or 12-digit dialing instead of
10, there will be an immense amount of agony.

So usually you don't see providers just handing out blocks of numbers
for free.

Perhaps this isn't the case in Germany.

DID wasn't invented to "get rid of number blocks".  DID was invented
to *allow* number blocks, by passing off the last digits of the dialled
number to a PBX.  Without that, a PBX (or attendant) has to answer the 
line and ask you for the extension you want.  That situation has the
virtue of only burning a single number, but is viewed as uglier.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[John Millican]

On Tuesday March 08 2005 2:58 pm, James Taylor wrote:
> Ok, used your sip.conf inbound works.  Outbound gets:
> "SIP/2.0 604 Does not exist anywhere"
>
> Any ideas?
> James
>
> On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou <[EMAIL PROTECTED]>
>
> wrote:
> > Yes it is working just fine for me with the same sip.conf that you have.
> > ??
> > Except the "permit=sip.broadvoice.com"
> > You can see my config at
> > http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
> >
> > Also what is your extensions.conf ?
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of James
> > Taylor
> > Sent: Tuesday, March 08, 2005 2:15 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
> > working
> >
> > Does anybody have Broadvoice outbound working?
> >
> > On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis <[EMAIL PROTECTED]>
> >

I have a same sip.conf and out and in are working well.  I have 
sip.broadvoice.com mapped to proxy.lax.broadvoice.com in my hosts file.  this 
is nice for me as i can use sip.broadvoice.com in all .conf and if i need to 
change the proxy i do so in the hosts file. I do not use ${EXTEN:1} in my 
outbound dial and i always dial 10 digits.
John Millican
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[Asterisk-Users] Broadvoice latest changes and still not working - solved HEYYY

[Jerry Geis]

Looks like I had mistyped that long password.
so the register statement was correct but the context was NOT correct 
off by 1 character in the middle.

I never susspected the password as it worked before the weekend changes. 
Thought it was OK.

Thanks to everyone..
Jerry
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RE: [Asterisk-Users] SIP - Call Park/Pickup and Canreinvite=yes at the same time??

[Eric_Doiron]

Sorry for the bad post.. corrections in
line in RED..

 

Sorry

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric_Doiron
Sent: Tuesday, March 08, 2005 3:25
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP -
Call Park/Pickup and Canreinvite=yes at thesame time??



 

Hi all,

 

I am trying to use canreinvite in sip.conf and park/pick up
calls at the same time. 

 

Problem:

When I have it set up so RTP goes through asterisk
(sip.conf: canreinvite=no),
# to xfer works fine. But, when I set it up so the RTP goes direct between
endpoints (sip.conf: canreinvite=yes) the # to xfer doesn’t
work. I believe this is because asterisk isn’t in the RTP path and the #
is being sent to the other phone instead.

 

I have read the wiki and followed the recommendations of
sip.conf: type=peer and extensions.conf cant contain a t or T in the dial
command.

 

Does anyone have this working? Or is it a known issue to not
be able to use these 2 features at the same time?

 

Comments and sample WORKING configs welcome!

 

Thanks,

-Eric






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[Asterisk-Users] DID in the U.S.

[Andreas Roedl]

Hello!

There is something I really don't get: As I ordered a PRI ISDN line in Germany 
with DID, I had not to pay anything for a "DID number block", now I'm trying 
to get a PRI ISDN in the U.S. (CA) and SBC wants to charge more than 200 
USD/month for numbers. I mean, this has nothing to do with DID, where 
everything that comes after the "base number" will be transmitted to the PBX 
anyway. Wasn't DID invented to get rid of number blocks?

Please enlighten me. Thanks.


Andi
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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

What is the output of the "show version"  ?

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 3:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working

Marios,

You are correct. Every phone on the system (8 or so) has a context=smvoice-sip
in the config for every phone.

This config was all working uptil last saturday when broadvoice made the 
changes.
It has not worked for outgoing calls since then. Incoming is still working.

This is one of my extensions.

[405]
type=friend
dtmfmode=rfc2833
username=405
secret=SECRET
disallow=all
allow=ulaw
allow=alaw
host=dynamic
context=smvoice-sip
insecure=very
callerid="Fred Smith" <405>


Jerry

--

Hmm!!

OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.

But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

 

-Original Message-
From: asterisk-users-bounces at lists.digium.com 

[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of Jerry
Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com 

Subject: [Asterisk-Users] Broadvoice latest changes and still not working

My extension.conf is below.

Jerry


-



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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Giudice, Salvatore]

Can you call anywhere or is this problem just with broadvoice? Is there
any type of firewall like a netscreen or iptables configured in your
setup, which may be blocking outbound UDP? Do you have a packet capture
of the traffic that is leaving your network for broadvoice? You should
try to inspect the signaling exchange with their proxy.

-Original Message-
From: Jerry Geis [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 08, 2005 2:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not
working

I tried removing the permit and that made no difference.

I can still call in to the box but no calls out.

Jerry

--

Yes it is working just fine for me with the same sip.conf that you have.
??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?

-Original Message-
From: asterisk-users-bounces at lists.digium.com

[mailto:asterisk-users-bounces at lists.digium.com
] On Behalf Of
James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
working

Does anybody have Broadvoice outbound working?

On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

>/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>

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Re: [Asterisk-Users] Please help with install *

[Victoria Alexandru]

As dean collins suggested I did:

make clean
make linux26
then make install

still got problems, this time with make linux26:

[EMAIL PROTECTED] zaptel]# make clean
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed 
zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core


[EMAIL PROTECTED] zaptel]# make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o
makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o
ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o
zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o
fxotune.c
cc -o fxotune fxotune.o -lm
make -C /lib/modules/`uname -r`/build
SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory
`/usr/src/linux-2.6.10-3mdk'
  CC [M]  /usr/src/zaptel/zaptel.o
In file included from include/linux/autoconf.h:1,
 from include/linux/config.h:4,
 from /usr/src/zaptel/zconfig.h:9,
 from /usr/src/zaptel/zaptel.c:40:
include/linux/rhconfig.h:43:194: operator '&&' has no
right operand
/usr/src/zaptel/zaptel.c: In function `zt_chan_write':
/usr/src/zaptel/zaptel.c:1726: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In function
`ioctl_load_zone':
/usr/src/zaptel/zaptel.c:2362: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In function
`zt_common_ioctl':
/usr/src/zaptel/zaptel.c:2714: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:2774: warning: ignoring
return value of `copy_to_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:2777: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:2859: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:2889: warning: ignoring
return value of `copy_to_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In function
`zt_chanandpseudo_ioctl':
/usr/src/zaptel/zaptel.c:3604: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:3614: warning: ignoring
return value of `copy_to_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:3617: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:3668: warning: ignoring
return value of `copy_to_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:3672: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: At top level:
/usr/src/zaptel/zaptel.c:167: warning: 'fcstab'
defined but not used
make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel] Error 2
make[1]: Leaving directory
`/usr/src/linux-2.6.10-3mdk'
make: *** [linux26] Error 2
[EMAIL PROTECTED] zaptel]# 

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Re: [Asterisk-Users] GotoIf problem

[Howard Lowndes]

On Wed, 2005-03-09 at 05:29, kurt x wrote:
>  I am trying to test how the GotoIf and $LEN functions work but am not
> succeeding is
> this venture.  When I dial and access voicemail with an ani of 3000
> the gotoif statement does not push the call to s|6.  Its goes through
> each line( 1,2,3,4,5,6,7) .  In additon when I dial with a 10 digit
> ani the s,3,Gotoif does not work.  It also goes through each line(
> 1,2,3,4,5,6,7)
> 
> Any help is greatly appreciated.

Have you tried removing the quotes?

> 
> Thanks
> 
> Kurt 
> 
> Asterisk CVS-HEAD-07/14/04-16:28:29 built by
> [EMAIL PROTECTED] on a i686 running Linux
> 
> 
> [globals]
> ${ext}=0
> SetGlobalVar(DIGITS=10)
> 
> 
> [vmail]
> exten => s,1,Answer
> exten => s,2,NoOp(${ext})
> exten => s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
> exten => s,4,GotoIf($[${CALLERIDNUM}  = "3000"]?s|6)
> exten => s,5,Voicemail(u${ext})
> exten => s,6,Background(pbx-invalid)
> exten => s,7,Hangup
> ___
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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[Asterisk-Users] Forwarded call flag

[Dr. Matthew Roller]

When I forward my PSTN phone(Qwest) to my cellphone and someone calls
it, my cellphone(AT&T) shows an arrow next to the caller id showing it
is a forwarded call, is there any way to set that forward flag when
forwarding a phone call through asterisk?

Its nice to know to answer a certain way when you are answering the
forwarded call
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Re: [Asterisk-Users] Wildcard X100P or TDM400P?

[Rich Adamson]

> On Tue, 2005-03-08 at 13:24 -0600, Rich Adamson wrote:
> > > > > TDM400P with FXO daughter card includes 1 hour of Digium support. It 
> > > > > is
> > > > > supposed to support other line types. If you have trouble, it is 
> > > > > likely
> > > > > you will get direct support from Digium and from the community here. 
> > > > 
> > > > It should be noted that several people including myself are having 
> > > > voicemail
> > > > volume problems with the TDM400P (is anyone having it with the X100P?) 
> > > > which
> > > > for us makes the card unusable for what was intended (a basic home 
> > > > PBX/answering
> > > > machine).
> > > > 
> > > > This is documented in bug #2023:
> > > > http://bugs.digium.com/bug_view_page.php?bug_id=0002023
> > > > 
> > > > You'll also notice that it has been dormant for quite some time.
> > > 
> > > Voicemail volume is not related to the card you use. If it was related
> > > to the card used, you wouldn't have the separation needed to make all
> > > the interfaces work.
> > > 
> > > Voicemail volume is usually complained about by people not using
> > > standard wav format. It is due to the volume of wav files being
> > > manipulated at write time where as all other formats are as they come
> > > off of the line.  
> > 
> > Steve, help me understand exactly what you said above. Having problems
> > with "to the card you use" and "related to the card used".
> 
> By the time any audio makes it to voicemail, it has been "normalized"
> into ast_frames. The source of audio is not relevant to the voicemail
> app.  
> 
> > Also, please clearify "not using standard wav format". Does that truly
> > mean recording voicemail messages in gsm format is the cause for the
> > additional 10db of loss measured and noted in bug 2023?
> 
> I don't know or care how you are measuring a 10db loss. I know for a
> fact that in format_wav.c there is a section of code that effectively
> doubles the volume as it is saving and removes the doubling on playback.
> No other audio format is given the same treatment.
> 
> > If that's the case, then why has bug 2023 been lurking without any such
> > comments for many many months?
> 
> If it is lurking for 10 months it is because no one cares to read it. I
> promise you I don't look at any bug unless it is mentioned on a list as
> is pertaining to code I am running. I have so little time now, I don't
> go looking for needles in haystacks that I don't need.

The only reason for questioning the above is that Mark (and several others)
have contributed various comments to bug 2022 and 2023, but at no time
have any of them ever mentioned comments relative to wav vs other file 
formats, and effectively have left the bug unresolved.

Rich


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[Asterisk-Users] SIP - Call Park/Pickup and Canreinvite=yes at the same time??

[Eric_Doiron]

Hi all,

 

I am trying to use canreinvite in sip.conf and park/pick up
calls at the same time. 

 

Problem:

When I have it set up so RTP goes through asterisk
(sip.conf: canreinvite=yes), # to xfer works fine. But, when I set it up so the
RTP goes direct between endpoints (sip.conf: canreinvite=no), the # to xfer doesn’t
work. I believe this is because asterisk isn’t in the RTP path and the #
is being sent to the other phone instead.

 

I have read the wiki and followed the recommendations of
sip.conf: type=peer and extensions.conf cant contain a t or T in the dial command.

 

Does anyone have this working? Or is it a known issue to not
be able to use these 2 features at the same time?

 

Comments and sample WORKING configs welcome!

 

Thanks,

-Eric






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[Asterisk-Users] Play music on hold while waiting for DTMF?

[David Brodbeck]

Is there a way to play music on hold for a specified amount of time while
listening for DTMF?  I suppose I'm looking for a hybrid of Background() and
WaitMusicOnHold().  I don't really want to use Background() because the
music would start over each time.

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Re: [Asterisk-Users] Options for Attendant Console.

[Kyle Hagan]

www.quadrasoftware.com
Kyle
Will McCown wrote:
We've been playing with Asterisk with an eye towards possibly
replacing or augmenting our existing PBX serving about over 600
phones (and needing to expand).  The one missing bit that I can't
find any mention of is an Attendant Console.  Are there any
good solutions out there?
I've considered that maybe one of the better softphones might
suffice, but the ones I've looked at so far not geared to the
need to handle and quickly dispatch large numbers of calls.
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Re: [Asterisk-Users] Cisco 7940 Upgrade Failing

[Rich Adamson]

> Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above.
> Can't get it to upgrade on its own via TFTP.  Phones w SCCP image will
> upgrade fine but I can't get these 2.0.3s to start the firmware upgrade.

On some older 79x0's, we've had to delete a bunch of the config statements
within the *.cnf files in order to upgrade. Deleting comments and the 
majority of statements won't have any impact in terms of upgrading, just
make sure you keep a full copy of the files around so that once the
upgrade is complete, you know what parameters were there to start with.

The above is based on upgrading several used 79x0's (with experience).
Seems not all phones were created equal (eg, maybe differences in buffer
sizes, nvram, or something).

I never did try to figure out why some worked and some did not without
deleting that stuff. 


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[Asterisk-Users] determining an available channel question

[Cirelle Internet Products]

Hello,
I have an outgoing dial plan which utilizes FWD for any of our
outgoing 800,877,866, etc while, toll and local calls get routed
to one of our pots lines.
when we use the pots lines, we use the chanisavail function
to choose an available pots line.
for example
exten => _1800NXX,1,ChanIsAvail(Zap/26&Zap/25)
exten => _1800NXX,2,Cut(theChannel=AVAILCHAN,,1)
exten => _1800NXX,3,Dial(${theChannel}/w${EXTEN}) 

works like a champ
Recently, we have had to stop using FWD because at times the call
cannot be processed (due to load?? probably). But there is a
voice response indicating the call cannot go through.
Currently, all tollfree calls chew up a pots line.
When this happens, I would like the call to failover to one of
the available pots lines.
ChanIsAvail is supposed to work with SIP from what I read, but
since FWD answers, the call is complete.
Has anybody confronted this type of configuration with any
success? Is it possible? If so, a pointer would be appreciated.
Best Regards
Greg

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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

Marios,
You are correct. Every phone on the system (8 or so) has a context=smvoice-sip
in the config for every phone.
This config was all working uptil last saturday when broadvoice made the 
changes.
It has not worked for outgoing calls since then. Incoming is still working.
This is one of my extensions.
[405]
type=friend
dtmfmode=rfc2833
username=405
secret=SECRET
disallow=all
allow=ulaw
allow=alaw
host=dynamic
context=smvoice-sip
insecure=very
callerid="Fred Smith" <405>
Jerry
--
Hmm!!
OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.
But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

-Original Message-
From: asterisk-users-bounces at lists.digium.com 
 
[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com 

Subject: [Asterisk-Users] Broadvoice latest changes and still not working
My extension.conf is below.
Jerry
-

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Re: [Asterisk-Users] Adit 600 for asterisk

[Steven Critchfield]

On Tue, 2005-03-08 at 14:07 -0600, Dennis Webb wrote:
> Ok,  I've pretty much decided to try the Adit route.  Somebody who has
> experience with these tell me if I'm missing something.
> 
> I have 15 incoming PSTN lines.  T1 is not an option at current
> location.  I want to put in an Adit 600 with 2 8-port FXO boards.  The
> adit will then connect to * via a digium t1 board.  I configure
> zaptel.conf for the T1.  What other parts would be needed?  How do the
> PSTN lines connect to the Adit, standard rj11 jacks? It looks to be
> about a $2500 investment and I need to know if there is anything
> special I am missing.

The Adit has a 50 pin D connector that is like the old SCSI connectors.
To connect your phone lines, you either want to get a punch down block
that has a 50 pin connector on it, or possibly a device with RJ11 style
jacks and a 50 pin connector. Then you connect it with a 25 pair cable
with 50 pin connectors on either side. 

Go to any reputable supplier near you and they should be able to help
you look at and find what you are comfortable with to use for
installation.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Please help with install *

[Steven Critchfield]

On Tue, 2005-03-08 at 11:53 -0800, Victoria Alexandru wrote:
> I started with a fresh Mandrake 10.2beta3 install, and
> bellow I'll show what I have on my system, ending with
> the attempt to compile zaptel. I failed, so I wouldn't
> go any further until I understand what is going wrong.
> Could anyone haelp please?

> Atempting to compile zaptel:
> [EMAIL PROTECTED] src]# cd zaptel
> [EMAIL PROTECTED] zaptel]# make clean ; make install 

> SUBDIRS=/usr/src/zaptel modules
> make[1]: Entering directory
> `/usr/src/linux-2.6.10-3mdk'
>   CC [M]  /usr/src/zaptel/zaptel.o
> In file included from include/linux/autoconf.h:1,
>  from include/linux/config.h:4,
>  from /usr/src/zaptel/zconfig.h:9,
>  from /usr/src/zaptel/zaptel.c:40:
> include/linux/rhconfig.h:43:194: operator '&&' has no
> right operand


Looks like a linux source code on your computer problem not a asterisk
and/or distribution incompatibility.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Broadvoice latest changes and still not working

[Marios Andreou]

Hmm!!

OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because 
incoming you said is working.

But what I don't know is the context for the SIP/ 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should "include => smvoice-sip"

 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working

My extension.conf is below.

Jerry


-


[default]
exten => s,1,Wait,1 ; Wait before speaking
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 
5 seconds
exten => s,4,ResponseTimeout,20 ; Set Response Timeout 
to 10 seconds
exten => s,5,ChanIsAvail(SIP/201&SIP/202&SIP/203&SIP/204&SIP/205&SIP/206)
exten => s,6,Cut(thechannel=AVAILCHAN,,1)
exten => s,7,Dial(${thechannel},${DIAL_TIMEOUT},tT)
exten => s,8,background(SM_ATTENDANT)
exten => s,9,noop("background done")
exten => s,10,SetVar(SMVOICE_EXTEN=${OPERATOR})
exten => s,11,Goto(default,operator,1)

exten => PHONE,1,Goto(default,s,1)


[smvoice-sip]
exten => 11,1,playback(demo-congrats)
exten => 11,2,hangup

exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)






Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?

-Original Message-
From: asterisk-users-bounces at lists.digium.com 

[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working

Does anybody have Broadvoice outbound working?

On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>

wrote:

>/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>/
/>/ ; Broadvoice
/>/ register => PHONE at sip.broadvoice.com  
/>/ :SECRET:PHONE  
/>/ at sip.broadvoice.com  
/>/ /PHONE
/>/
/>/ [Broadvoice]
/>/ type=friend
/>/ username=PHONE
/>/ authuser=PHONE
/>/ fromuser=PHONE
/>/ secret=secret
/>/ host=sip.broadvoice.com
/>/ port=5060
/>/ context=default
/>/ fromdomain=sip.broadvoice.com
/>/ canreinvite=no
/>/ dtmfmode=inband
/>/ insecure=very
/>/ permit=sip.broadvoice.com
/>/ qualify=yes
/>/ disallow=all
/>/ allow=ulaw
/>/ maxexpirey=180
/>/ defaultexpirey=160
/>/ videosupport=no
/>/
/>/
/>/ exten =>  
/>/ _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ exten =>  
/>/ 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ ___
/>/ Asterisk-Users mailing list
/>/ Asterisk-Users at lists.digium.com 

/>/ http://lists.digium.com/mailman/listinfo/asterisk-users
/>/ To UNSUBSCRIBE or update options visit:
/>/http://lists.digium.com/mailman/listinfo/asterisk-users
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/


-- 
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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  still not working
  
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Re: [Asterisk-Users] Wildcard X100P or TDM400P?

[Steven Critchfield]

On Tue, 2005-03-08 at 13:24 -0600, Rich Adamson wrote:
> > > > TDM400P with FXO daughter card includes 1 hour of Digium support. It is
> > > > supposed to support other line types. If you have trouble, it is likely
> > > > you will get direct support from Digium and from the community here. 
> > > 
> > > It should be noted that several people including myself are having 
> > > voicemail
> > > volume problems with the TDM400P (is anyone having it with the X100P?) 
> > > which
> > > for us makes the card unusable for what was intended (a basic home 
> > > PBX/answering
> > > machine).
> > > 
> > > This is documented in bug #2023:
> > > http://bugs.digium.com/bug_view_page.php?bug_id=0002023
> > > 
> > > You'll also notice that it has been dormant for quite some time.
> > 
> > Voicemail volume is not related to the card you use. If it was related
> > to the card used, you wouldn't have the separation needed to make all
> > the interfaces work.
> > 
> > Voicemail volume is usually complained about by people not using
> > standard wav format. It is due to the volume of wav files being
> > manipulated at write time where as all other formats are as they come
> > off of the line.  
> 
> Steve, help me understand exactly what you said above. Having problems
> with "to the card you use" and "related to the card used".

By the time any audio makes it to voicemail, it has been "normalized"
into ast_frames. The source of audio is not relevant to the voicemail
app.  

> Also, please clearify "not using standard wav format". Does that truly
> mean recording voicemail messages in gsm format is the cause for the
> additional 10db of loss measured and noted in bug 2023?

I don't know or care how you are measuring a 10db loss. I know for a
fact that in format_wav.c there is a section of code that effectively
doubles the volume as it is saving and removes the doubling on playback.
No other audio format is given the same treatment.

> If that's the case, then why has bug 2023 been lurking without any such
> comments for many many months?

If it is lurking for 10 months it is because no one cares to read it. I
promise you I don't look at any bug unless it is mentioned on a list as
is pertaining to code I am running. I have so little time now, I don't
go looking for needles in haystacks that I don't need.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Broadvoice latest changes and still not working

[Roger Hanson]

My broadvoice works perfectly.  I am using a standard registration 
string,  however.  Not the funky one broadvoice says to use.  I can make 
outbound and receive inbound calls over broadvoice.

I'm using AMP also.
register=phonenumber:[EMAIL PROTECTED]
sip.conf:
[952XX]
username=952XX
type=friend
secret=password
regexten=952XXX
insecure=very
host=sip.broadvoice.com
fromuser=952XX
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes
authuser=952
[sdfdsf]
- Original Message - 
From: "Marios Andreou" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Sent: Tuesday, March 08, 2005 1:18 PM
Subject: RE: [Asterisk-Users] Broadvoice latest changes and still not 
working


Yes it is working just fine for me with the same sip.conf that you 
have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html

Also what is your extensions.conf ?
-Original Message-
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[Asterisk-Users] Adit 600 for asterisk

[Dennis Webb]

Ok,  I've pretty much decided to try the Adit route.  Somebody who has experience with these tell me if I'm missing something.

I have 15 incoming PSTN lines.  T1 is not an option at current location.  I want to put in an Adit 600 with 2 8-port FXO boards.  The adit will then connect to * via a digium t1 board.  I configure zaptel.conf for the T1.  What other parts would be needed?  How do the PSTN lines connect to the Adit, standard rj11 jacks? It looks to be about a $2500 investment and I need to know if there is anything special I am missing.

Thanks.


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[Asterisk-Users] Oooh, 'SIP/caroline-cc80' was 10.44.100.1:8000/(format 1544)

[Iqbal]

Hi

I keep getting this in my debug, and the next line usually reads

 Oooh, 'SIP/caroline-cc80' wasv 0.0.0.0:0/(format 1544)

caroline is a context I have defined for incoming calls via pstn --> ser
---> asterisk for a locally logged user.

But when I dial from a pstn line the call goes through but no voice, I
wanted to know if its something to do with the message above

Iqbal
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RE: [Asterisk-Users] GotoIf with Authenticate

[Guy C. Guckenberger]

That is just a typeO in the 
email:
 
 
Just to be clear...Im trying to pull to 
extension off the IP phone and evaluate it against 2006.  Is EXTEN the 
correct variable?
 
 
 
 

exten => s,1,GotoIf($[${EXTEN} = "2006"]?3)
exten => s,2,Authenticate(731)
exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) 
exten => s,4,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,5,Goto(6)
exten => s,6,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) 
exten => s,7,SetCallerID(${OUTCID_${ARG1}})
exten => s,8,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten => s,9,Dial(${OUT_${ARG1}}/w${ARG2:${length}})
exten => s,10,Congestion
exten => s,108,Macro(outisbusy)
 
 


From: [EMAIL PROTECTED] on 
behalf of John FullingtonSent: Tue 3/8/2005 2:55 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] GotoIf with Authenticate

Change 
extex in the first line to exten
 
 -Original Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Guy C. 
GuckenbergerSent: Tuesday, March 08, 2005 2:46 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] GotoIf with 
Authenticate

  Quick question...Im authenticate 
  all exten except this one(2006). If I call from ext 2006 I still have to 
  authenticate.  If I call form any other ext I have to authenticate. 
   Any suggestions?
   
   
  Thanks
   
   
  
  extex => s,1,GotoIf($[${EXTEN} = "2006"]?3)
  exten => s,2,Authenticate(731)
  exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) 
  exten => s,4,SetCallerID(${ECID${CALLERIDNUM}})
  exten => s,5,Goto(6)
  exten => s,6,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) 
  exten => s,7,SetCallerID(${OUTCID_${ARG1}})
  exten => s,8,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
  exten => s,9,Dial(${OUT_${ARG1}}/w${ARG2:${length}})
  exten => s,10,Congestion
  exten => 
s,108,Macro(outisbusy)
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[Asterisk-Users] Cisco 7940 Upgrade Failing

[Staalenburg, Juan]

Does anyone know how to get a Cisco 7940 w/FW ver 2.0.3 to v3x and above.
Can't get it to upgrade on its own via TFTP.  Phones w SCCP image will
upgrade fine but I can't get these 2.0.3s to start the firmware upgrade.

Thanks.


Regards,

Juan Staalenburg
Teksavers, Inc.
(512) 255-8395 x1002
AIM: juanteksavers

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RE: [Asterisk-Users] GotoIf with Authenticate

[John Fullington]

Change extex in the first line to exten
 
 -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Guy C.
Guckenberger
Sent: Tuesday, March 08, 2005 2:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GotoIf with Authenticate



Quick question...Im authenticate all exten except this one(2006). If I call
from ext 2006 I still have to authenticate.  If I call form any other ext I
have to authenticate.  Any suggestions?
 
 
Thanks
 
 
extex => s,1,GotoIf($[${EXTEN} = "2006"]?3)

exten => s,2,Authenticate(731)

exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) 

exten => s,4,SetCallerID(${ECID${CALLERIDNUM}})

exten => s,5,Goto(6)

exten => s,6,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) 

exten => s,7,SetCallerID(${OUTCID_${ARG1}})

exten => s,8,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})

exten => s,9,Dial(${OUT_${ARG1}}/w${ARG2:${length}})

exten => s,10,Congestion

exten => s,108,Macro(outisbusy)

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Re: [Asterisk-Users] Please help with install *

[Victoria Alexandru]

I started with a fresh Mandrake 10.2beta3 install, and
bellow I'll show what I have on my system, ending with
the attempt to compile zaptel. I failed, so I wouldn't
go any further until I understand what is going wrong.
Could anyone haelp please?

Here is the kernel I'm running:
[EMAIL PROTECTED] victoria]# uname -r
2.6.10-3mdk

Required packages and install status:

- kernel sources:
[EMAIL PROTECTED] victoria]# rpm -qa |grep kernel
kernel-source-2.6-2.6.10-3mdk
kernel-2.6.10.3mdk-1-1mdk

- ncurses and associated -devel:
[EMAIL PROTECTED] victoria]# rpm -qa |grep ncurses
libncurses5-5.4-1.20050108.1mdk
libncurses5-devel-5.4-1.20050108.1mdk
ncurses-5.4-1.20050108.1mdk

- openssl and associated -devel:
[EMAIL PROTECTED] victoria]# rpm -qa |grep openssl
libopenssl0.9.7-static-devel-0.9.7e-5mdk
libopenssl0.9.7-devel-0.9.7e-5mdk
openssl-0.9.7e-5mdk
libopenssl0.9.7-0.9.7e-5mdk

- zlib and associated -devel:
[EMAIL PROTECTED] victoria]# rpm -qa |grep zlib
zlib1-devel-1.2.2.2-2mdk
zlib1-1.2.2.2-2mdk

- bison and associated -devel: I couldn't find a
-devel package :(
[EMAIL PROTECTED] victoria]# urpmi -y bison
The following packages contain bison:
bison
bison++

All bison related packages I have installed:
[EMAIL PROTECTED] victoria]# rpm -qa |grep bison
bison-2.0-1mdk

Regarding mpg123 I installed it from CD:
[EMAIL PROTECTED] victoria]# rpm -qa |grep mpg123
mpg123-0.59r-23mdk

Checking out from CVS:
[EMAIL PROTECTED] victoria]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] src]# cvs login
Logging in to
:pserver:[EMAIL PROTECTED]:2401/usr/cvsroot
CVS password:
[EMAIL PROTECTED] src]# cvs checkout zaptel libpri
asterisk asterisk-addons asterisk-sounds
...
U asterisk-sounds/sounds/wx/temperature.gsm
U asterisk-sounds/sounds/wx/wind-chill.gsm
U asterisk-sounds/sounds/wx/winds.gsm
[EMAIL PROTECTED] src]#

Atempting to compile zaptel:
[EMAIL PROTECTED] src]# cd zaptel
[EMAIL PROTECTED] zaptel]# make clean ; make install 
rm -f torisatool makefw tor2fw.h radfw.h
rm -f ztcfg torisatool makefw ztmonitor ztspeed 
zttest fxotune
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f *.ko *.mod.c .*o.cmd
rm -rf .tmp_versions
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f fxotune
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c   -o
makefw
./makefw tormenta2.rbt tor2fw > tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw > radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o ztcfg.o
ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"
-DBUILDING_TONEZONE -o zonedata.lo zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"
-DBUILDING_TONEZONE -o tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o -lm -L. libtonezone.a
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o
ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"zttest.c   -o
zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o
fxotune.c
cc -o fxotune fxotune.o -lm
make -C /lib/modules/`uname -r`/build
SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory
`/usr/src/linux-2.6.10-3mdk'
  CC [M]  /usr/src/zaptel/zaptel.o
In file included from include/linux/autoconf.h:1,
 from include/linux/config.h:4,
 from /usr/src/zaptel/zconfig.h:9,
 from /usr/src/zaptel/zaptel.c:40:
include/linux/rhconfig.h:43:194: operator '&&' has no
right operand
/usr/src/zaptel/zaptel.c: In function `zt_chan_write':
/usr/src/zaptel/zaptel.c:1726: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In function
`ioctl_load_zone':
/usr/src/zaptel/zaptel.c:2362: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c: In function
`zt_common_ioctl':
/usr/src/zaptel/zaptel.c:2714: warning: ignoring
return value of `copy_from_user', declared with
attribute warn_unused_result
/usr/src/zaptel/zaptel.c:2774: warning: ignoring
return value of `copy_to_use

[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

I tried removing the permit and that made no difference.
I can still call in to the box but no calls out.
Jerry
--
Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: asterisk-users-bounces at lists.digium.com 
 
[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>
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[Asterisk-Users] GotoIf with Authenticate

[Guy C. Guckenberger]

Quick question...Im authenticate all exten except this 
one(2006). If I call from ext 2006 I still have to authenticate.  If I 
call form any other ext I have to authenticate.  Any 
suggestions?
 
 
Thanks
 
 

extex => s,1,GotoIf($[${EXTEN} = "2006"]?3)
exten => s,2,Authenticate(731)
exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) 
exten => s,4,SetCallerID(${ECID${CALLERIDNUM}})
exten => s,5,Goto(6)
exten => s,6,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6) 
exten => s,7,SetCallerID(${OUTCID_${ARG1}})
exten => s,8,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten => s,9,Dial(${OUT_${ARG1}}/w${ARG2:${length}})
exten => s,10,Congestion
exten => s,108,Macro(outisbusy)___
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RE: [Asterisk-Users] DTMF out to Cell Phone

[John Fullington]

Asterisk is calling out on a Zap channel, using a pri line.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:59 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] DTMF out to Cell Phone


Normally a problem with either reinvite or your dtmf type (info, inband,
etc).
Change these until it works.

On Tue, 8 Mar 2005 14:16:18 -0500, John Fullington <[EMAIL PROTECTED]> wrote:

> I set up a monitoring system that calls my techs when a problem occurs on
> one of our networks, everything works fine unless  asterisk calls a cell
> phone in which case the tech can not respond using dtmf. It works fine if
> the tech call in but not if asterisk call a tech's cell phone. Anyone one
> have any suggestions?
>
> Thanks
> John Fullington
>
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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Spam detection software, running on the system "zeus.avanzada7.com", has
identified this incoming email as possible spam.  The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email.  If you have any questions, see
the administrator of that system for details.

Content preview:  Normally a problem with either reinvite or your dtmf
  type (info, inband, etc). Change these until it works. On Tue, 8 Mar
  2005 14:16:18 -0500, John Fullington <[EMAIL PROTECTED]> wrote: > I set
  up a monitoring system that calls my techs when a problem occurs on >
  one of our networks, everything works fine unless asterisk calls a
  cell > phone in which case the tech can not respond using dtmf. It
  works fine if > the tech call in but not if asterisk call a tech's
  cell phone. Anyone one > have any suggestions? > > Thanks > John
  Fullington > > > Asterisk-Users mailing list >
  Asterisk-Users@lists.digium.com >
  http://lists.digium.com/mailman/listinfo/asterisk-users > To
  UNSUBSCRIBE or update options visit: >
  http://lists.digium.com/mailman/listinfo/asterisk-users > [...]

Content analysis details:   (0.1 points, 5.0 required)

 pts rule name  description
 -- 
--
 0.1 FORGED_RCVD_HELO   Received: contains a forged HELO

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[Asterisk-Users] Broadvoice latest changes and still not working

[Jerry Geis]

My extension.conf is below.
Jerry
-
[default]
exten => s,1,Wait,1 ; Wait before speaking
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 
5 seconds
exten => s,4,ResponseTimeout,20 ; Set Response Timeout 
to 10 seconds
exten => s,5,ChanIsAvail(SIP/201&SIP/202&SIP/203&SIP/204&SIP/205&SIP/206)
exten => s,6,Cut(thechannel=AVAILCHAN,,1)
exten => s,7,Dial(${thechannel},${DIAL_TIMEOUT},tT)
exten => s,8,background(SM_ATTENDANT)
exten => s,9,noop("background done")
exten => s,10,SetVar(SMVOICE_EXTEN=${OPERATOR})
exten => s,11,Goto(default,operator,1)
exten => PHONE,1,Goto(default,s,1)
[smvoice-sip]
exten => 11,1,playback(demo-congrats)
exten => 11,2,hangup
exten => 
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten => 
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)


Yes it is working just fine for me with the same sip.conf that you have. ??
Except the "permit=sip.broadvoice.com"
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: asterisk-users-bounces at lists.digium.com 
 
[mailto:asterisk-users-bounces at lists.digium.com 
] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>>  
wrote:

/ Here is my configs. from a previous post...
/>/
/>/ Jerry
/>/
/>/ --
/>/
/>/ ; Broadvoice
/>/ register => PHONE at sip.broadvoice.com  
/>/ :SECRET:PHONE  
/>/ at sip.broadvoice.com  
/>/ /PHONE
/>/
/>/ [Broadvoice]
/>/ type=friend
/>/ username=PHONE
/>/ authuser=PHONE
/>/ fromuser=PHONE
/>/ secret=secret
/>/ host=sip.broadvoice.com
/>/ port=5060
/>/ context=default
/>/ fromdomain=sip.broadvoice.com
/>/ canreinvite=no
/>/ dtmfmode=inband
/>/ insecure=very
/>/ permit=sip.broadvoice.com
/>/ qualify=yes
/>/ disallow=all
/>/ allow=ulaw
/>/ maxexpirey=180
/>/ defaultexpirey=160
/>/ videosupport=no
/>/
/>/
/>/ exten =>  
/>/ _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ exten =>  
/>/ _91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
/>/
/>/ ___
/>/ Asterisk-Users mailing list
/>/ Asterisk-Users at lists.digium.com 
/>/ http://lists.digium.com/mailman/listinfo/asterisk-users
/>/ To UNSUBSCRIBE or update options visit:
/>/http://lists.digium.com/mailman/listinfo/asterisk-users
/>/
/

--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas  75503
903-793-1956
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