from:"VoIP"

Hello all,

We would like to inform the caller of the reason for a failed call.

For example, when we get a 486 Busy Here, the system accepts it and in the
CLI we see Everyone is busy/congested at this time.

Can we use this data to play an announcement to the caller?

Thank you in advance for your help.

Michael
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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

Thank you Kevin,

We'll upgrade our server to 1.6.2.12 and try again.

Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
extensions that our SIP provider gives us to a fax storage server or later
to email.

Michael

On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.comwrote:


 You have a 'Local' channel in between SendFAX and the SIP channel to
 your other endpoint. In Asterisk 1.6.2.11, chan_local was not properly
 aware of T.38 negotiation, so it ends up acting as a sort of 'firewall'
 between the endpoints.

 This was fixed in Asterisk 1.6.2.12 and later releases, so if you were
 running the current version, you wouldn't have experienced this specific
 problem. This was listed in the ChangeLog for 1.6.2.12, but
 unfortunately the commit message the developer wrote did not explain why
 the change was made or what problem it was addressing, so you wouldn't
 have noticed it.

 In any case, upgrading to 1.6.2.12 or later will cure this problem.


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[asterisk-users] 2 step dialing

Hello all,

We're trying to build a small IVR application to allow callers to use the
Asterisk for outgoing calls in a 2 steps dialing mode.

The context for outgoing calls is called outgoing (we have there an LCR
and routing mechanism we want to use, depending on the destination).

This is what we did, but it doesn't work:
exten = _X., 13, Read(ccdest,vm-enter-num-to-call,,,2)
exten = _X., 14, NoOp($ccdest)
exten = _X., 15, Dial(Local/$ccd...@outgoing,50)

The error we get is:
chan_local.c:538 local_call: No such extension/context
$ccd...@outgoingwhile calling Local channel
-- Couldn't call $ccd...@outgoing

We know there's a syntax problem in line 15, but not sure how to fix it.

Thank you for your help.

Michael
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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

Hello again,

If I set a peer to use G.711 only, they try to process a sent fax in G.711,
but Asterisk doesn't like it:

WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on
channel 'SIP/Main-000a' and T.38 negotiation failed; aborting.

What can I do to enable it?

Thanks,

Michael
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[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

   Hello,

I'm trying to send a tif file, using Fax for Asterisk and the call is 
executed, but when I get the reINVITE with T.38 data, the local server 
doesn't recognize that we have this capability and sends a 488 message. 
These are the logs:

--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
INVITE sip:1234...@10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb
To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5
Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, 
application/dtmf-relay,  multipart/mixed
Contact: sip:98765...@xxx.xxx.xxx.xx8:5060
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length:  303
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6202 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

-
--- (16 headers 13 lines) ---
Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 
74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 
(nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8
From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb
To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5
Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:1234...@yyy.yyy.yyy.yyy
Content-Length: 0




--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8
From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb
To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5
Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0



Please help.

Thank you.

Michael



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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

It's set to yes for this peer.

also t38pt_udptl is set to yes.

:(

On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote:

 On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com
 wrote:
Hello,
 
  I'm trying to send a tif file, using Fax for Asterisk and the call is
  executed, but when I get the reINVITE with T.38 data, the local server
  doesn't recognize that we have this capability and sends a 488 message.

 http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

 take a look at your canreinvite option.

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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

We don't have an ATA and fax machine.

The whole point (as I specified in the header and initial message) is the
attempt to use Fax for Asterisk to send the message.

As I showed in the logs, the remote carrier sends a proper T.38 reINVITE,
but our Asterisk doesn't accept, despite the fact that this provider is
defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the
only question is (as far as we understand) is why in this scenario, the T.38
is rejected.

Here are the logs (sip debug is open) again, since we get the reINVITE:

--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
INVITE sip:1234...@10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
From: 
sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb

To: Fax sip:1234...@yyy.yyy.yyy.yyy
sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay,  multipart/mixed
Contact: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060
Supported: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Length:  303
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8
s=SIP Media Capabilities
c=IN IP4 xxx.xxx.xxx.xx7
t=0 0
m=image 6202 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

-
--- (16 headers 13 lines) ---
Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT)
Got T.38 offer in SDP in dialog
74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8

From: 
sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb

To: Fax sip:1234...@yyy.yyy.yyy.yyy
sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy
Content-Length: 0




--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8

From: 
sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb

To: Fax sip:1234...@yyy.yyy.yyy.yyy
sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5

Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Server: Smartel-PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Thanks.

Michael


On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg dbackeb...@gmail.comwrote:

 On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com
 wrote:
  It's set to yes for this peer.
 
  also t38pt_udptl is set to yes.
 
  :(

 You don't say anything about what you're trying to send / receive against.

 Here's how you should troubleshoot:

 * start with a 'real fax machine' if you have one, on an analog line
 if you have one. If you can't receive / send with that against your
 target, blame your target.
 * move to audio-pass through fax on asterisk. No T.38. If that works.
 * add in T.38

 You will learn things in that process and be able to tell at what
 layer your troubles are happening.

 It could be coincidental that things give up during the reinvite. It
 could actually be giving up for noise on the line, packet drops, etc.

 At the very least, start recording the call. You'll at least be able
 to hear up to the re-invite.

 Definitely record the audio passthrough attempt and listen back to it.

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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test one
doesn't work?

I know they read (and sometimes respond) to this list, so I don't understand
why they don't clarify this issue.

I spent a few hours on Google and saw many similar posts, but no actual
valuable answer.

Weird...

On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas da...@debsinc.com wrote:

   From what I have read over the last few months, you should invest in
 Motrin before trying T.38 faxing with or without FFA – it can (possibly) be
 done, but it has beaten some folks into the ground trying it.



 Could be a codec issue.

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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)

: 8530 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0



sip*CLI
--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
ACK sip:98765...@10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a
From: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef
To: Fax sip:98765...@yyy.yyy.yyy.yyy;tag=as28606a47
Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy
CSeq: 8530 ACK
Max-Forwards: 70
Content-Length: 0



Thanks,

Michael


On Tue, Oct 19, 2010 at 8:56 PM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 10/19/2010 12:01 PM, VoIP Question wrote:
  Digium claims that their FFA is the best and most compatible solution
  and they give one channel for free, but do not provide support for those
  that do not buy more channels, but why buy more channels if the
  free/test one doesn't work?
 
  I know they read (and sometimes respond) to this list, so I don't
  understand why they don't clarify this issue.

 When you are asking for free help on a mailing list, patience is a
 virtue :-) You posted your question approximately four hours ago.

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 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Checking SIP Headers existence and content

2010-10-04 Thread VoIP Question

Hello,

I would like to verify if a specific SIP header exists, and if yes, extract
the partial content from another header.

1. Is there a way to verify if a specific header exists?
2. How do I extract data that is between the first : and the following @?
Specifically, The data looks like sip:1234567...@10.0.0.1:5060 and I would
like to get only the 1234567890

I tried to use the CUT() command, but without success.

Thank you.

Michael
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[asterisk-users] Switchboad like application

2010-06-21 Thread voip crazy

Hello all,

Anybody could point me any clue about an Open Source or licensed
switchboard for my users?
ARI or FOP is not enought for my users.

Thanks in advance.

VoipCrazy

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Re: [asterisk-users] Qwest PRIs

2010-06-16 Thread Voip Asterisk

Ok got it up and running. In the case for Qwest with NFAS they reserve what
they call Interface ID 1 for the circuit with the backup d channel. In
our case we only have two circuits with a single d channel. The real key
was realizing the logical span number in the spanmap translated into
interface ID so here are the spanmaps that worked for us:

[trunkgroups]
trunkgroup = 1,24
spanmap = 1,1,0
spanmap = 2,1,2

group=1
switchtype=dms100
echocancel=yes
signalling=pri_cpe
channel =1-23,25-48

Notice the switchtype. While they told me that their switchtype was NI2
(National ISDN 2) they did say they were using a dms100, so I would always
ask your carrier what switch they have.

Anyway thanks for your help all.

On Mon, Jun 14, 2010 at 4:47 PM, Steve Edwards asterisk@sedwards.comwrote:

On Mon, 14 Jun 2010, C F wrote:

One more thing, read the comments here:

http://www.voip-info.org/wiki/index.php?page_id=573tk=2ff846f8169b7694aed5comments_page=1
Don't forget to have a beer ready :P

Now that's really funny.

I read along with this and was thinking this was exactly my experience
with some Qwest PRIs a couple of years ago.

Then I noticed -- it was me :)

I guess I had too many beers.

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Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Qwest PRIs

2010-06-14 Thread Voip Asterisk

Ya I'm passed that part now.  I have dahdi properly loading the card, and
both links are green.  Asterisk recognizes the channels, but still shows the
span as down.

On Sun, Jun 13, 2010 at 6:35 PM, C F shma...@gmail.com wrote:

 Not sure what version you are running but I'm still running 1.2x in
 1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
 assuming layer 2+) part loads with Asterisk.

 On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk aster...@wideideas.com
 wrote:
  Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi
 to
  bring up the PRIs without alarms.
 
  On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle supp...@drdos.info wrote:
 
  Voip Asterisk wrote:
   Hi,
  
   I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
I'm using an OpenVox D410E and the drivers are loaded.  My
   system.conf looks like this:
  
 
  Google turned up this:
 
  http://www.voip-info.org/wiki/view/NFAS
 
  Doug
 
  --
  Ben Franklin quote:
 
  Those who would give up Essential Liberty to purchase a little
 Temporary
  Safety, deserve neither Liberty nor Safety.
 
 
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Re: [asterisk-users] Qwest PRIs

2010-06-13 Thread Voip Asterisk

Ok making a little progress, but still can't get them completely up:

 pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3
Overlap Recv: No


cat /proc/dahdi/1; cat /proc/dahdi/2
Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF
CRC4 error count: 1293
E-bit error count: 2774

   1 TE4/0/1/1 Clear (In use)
   2 TE4/0/1/2 Clear (In use)
   3 TE4/0/1/3 Clear (In use)
   4 TE4/0/1/4 Clear (In use)
   5 TE4/0/1/5 Clear (In use)
   6 TE4/0/1/6 Clear (In use)
   7 TE4/0/1/7 Clear (In use)
   8 TE4/0/1/8 Clear (In use)
   9 TE4/0/1/9 Clear (In use)
  10 TE4/0/1/10 Clear (In use)
  11 TE4/0/1/11 Clear (In use)
  12 TE4/0/1/12 Clear (In use)
  13 TE4/0/1/13 Clear (In use)
  14 TE4/0/1/14 Clear (In use)
  15 TE4/0/1/15 Clear (In use)
  16 TE4/0/1/16 Clear (In use)
  17 TE4/0/1/17 Clear (In use)
  18 TE4/0/1/18 Clear (In use)
  19 TE4/0/1/19 Clear (In use)
  20 TE4/0/1/20 Clear (In use)
  21 TE4/0/1/21 Clear (In use)
  22 TE4/0/1/22 Clear (In use)
  23 TE4/0/1/23 Clear (In use)
  24 TE4/0/1/24 HDLCFCS (In use)
Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF

  25 TE4/0/2/1 Clear (In use)
  26 TE4/0/2/2 Clear (In use)
  27 TE4/0/2/3 Clear (In use)
  28 TE4/0/2/4 Clear (In use)
  29 TE4/0/2/5 Clear (In use)
  30 TE4/0/2/6 Clear (In use)
  31 TE4/0/2/7 Clear (In use)
  32 TE4/0/2/8 Clear (In use)
  33 TE4/0/2/9 Clear (In use)
  34 TE4/0/2/10 Clear (In use)
  35 TE4/0/2/11 Clear (In use)
  36 TE4/0/2/12 Clear (In use)
  37 TE4/0/2/13 Clear (In use)
  38 TE4/0/2/14 Clear (In use)
  39 TE4/0/2/15 Clear (In use)
  40 TE4/0/2/16 Clear (In use)
  41 TE4/0/2/17 Clear (In use)
  42 TE4/0/2/18 Clear (In use)
  43 TE4/0/2/19 Clear (In use)
  44 TE4/0/2/20 Clear (In use)
  45 TE4/0/2/21 Clear (In use)
  46 TE4/0/2/22 Clear (In use)
  47 TE4/0/2/23 Clear (In use)
  48 TE4/0/2/24 Clear (In use)

cat system.conf | grep -v #
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24

span=2,2,1,esf,b8zs
bchan=25-48
loadzone= us
defaultzone = us

cat chan_dahdi.conf

[trunkgroups]
trunkgroup = 1,24
spanmap = 1,1,1
spanmap = 2,1,2

[channels]
context=incoming
switchtype=national
facilityenable=no
signalling=pri_cpe
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
channel = 1-23

group=2
channel = 25-48

If anyone could let me know what I should be doing next.  I'm sure my issue
is:

Status: Provisioned, Down, Active

specifically the Down part.

Thanks

On Sat, Jun 12, 2010 at 9:04 AM, Voip Asterisk aster...@wideideas.comwrote:

 BTW these were just up and running on a Cisco AS5300, so we know they work
 and Qwest has them turned up and configured correctly.  The only settings
 that were needed on the cisco was the coding and framing which is how dahdi
 is configured now.  Anyone have any idea how to mimic default cisco settings
 on dahdi/asterisk?

 The line card used in the cisco was the standard 4 port T1 PRI card.

 Thanks

 On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk aster...@wideideas.comwrote:

 Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi
 to bring up the PRIs without alarms.


 On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle supp...@drdos.info wrote:

 Voip Asterisk wrote:
  Hi,
 
  I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
   I'm using an OpenVox D410E and the drivers are loaded.  My
  system.conf looks like this:
 

 Google turned up this:

 http://www.voip-info.org/wiki/view/NFAS

 Doug

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 Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Qwest PRIs

2010-06-13 Thread Voip Asterisk

Ya 99% sure that isn't it since they were just pulled working off an AS5300

On Sun, Jun 13, 2010 at 4:27 AM, Doug Lytle supp...@drdos.info wrote:

 Voip Asterisk wrote:
 
  Status: Provisioned, Down, Active
 
  specifically the Down part.
 


 In my experience that usually means the provider hasn't brought up the
 PRI.  Granted, I've never used anything beyond a single PRI.

 Doug


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[asterisk-users] Qwest PRIs

2010-06-12 Thread Voip Asterisk

Hi,

I'm trying to bring up two PRIs from qwest with asterisk and dahdi.  I'm
using an OpenVox D410E and the drivers are loaded.  My system.conf looks
like this:

# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF RED
span=1,2,0,esf,b8zs
bchan=1-24

# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 (MASTER) B8ZS/ESF RED
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

These are suppose to be configured with NFAS (Non-Facilities Associated
Signaling) so 1 d channels manages all the b channels in both spans.  I
can't seem to get it out of alarm RED, asterisk running or not doesn't make
a difference.

 cat /proc/dahdi/1*;cat /proc/dahdi/2*
Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF RED

   1 TE4/0/1/1 Clear RED
   2 TE4/0/1/2 Clear RED
   3 TE4/0/1/3 Clear RED
   4 TE4/0/1/4 Clear RED
   5 TE4/0/1/5 Clear RED
   6 TE4/0/1/6 Clear RED
   7 TE4/0/1/7 Clear RED
   8 TE4/0/1/8 Clear RED
   9 TE4/0/1/9 Clear RED
  10 TE4/0/1/10 Clear RED
  11 TE4/0/1/11 Clear RED
  12 TE4/0/1/12 Clear RED
  13 TE4/0/1/13 Clear RED
  14 TE4/0/1/14 Clear RED
  15 TE4/0/1/15 Clear RED
  16 TE4/0/1/16 Clear RED
  17 TE4/0/1/17 Clear RED
  18 TE4/0/1/18 Clear RED
  19 TE4/0/1/19 Clear RED
  20 TE4/0/1/20 Clear RED
  21 TE4/0/1/21 Clear RED
  22 TE4/0/1/22 Clear RED
  23 TE4/0/1/23 Clear RED
  24 TE4/0/1/24 Clear RED
Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 (MASTER) B8ZS/ESF RED

  25 TE4/0/2/1 Clear RED
  26 TE4/0/2/2 Clear RED
  27 TE4/0/2/3 Clear RED
  28 TE4/0/2/4 Clear RED
  29 TE4/0/2/5 Clear RED
  30 TE4/0/2/6 Clear RED
  31 TE4/0/2/7 Clear RED
  32 TE4/0/2/8 Clear RED
  33 TE4/0/2/9 Clear RED
  34 TE4/0/2/10 Clear RED
  35 TE4/0/2/11 Clear RED
  36 TE4/0/2/12 Clear RED
  37 TE4/0/2/13 Clear RED
  38 TE4/0/2/14 Clear RED
  39 TE4/0/2/15 Clear RED
  40 TE4/0/2/16 Clear RED
  41 TE4/0/2/17 Clear RED
  42 TE4/0/2/18 Clear RED
  43 TE4/0/2/19 Clear RED
  44 TE4/0/2/20 Clear RED
  45 TE4/0/2/21 Clear RED
  46 TE4/0/2/22 Clear RED
  47 TE4/0/2/23 Clear RED
  48 TE4/0/2/24 HDLCFCS RED

Any ideas why these are alarmed?  Also what settings need to be set in
chan_dahdi.conf for asterisk to bring these up?

Thanks

Miles
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Re: [asterisk-users] Qwest PRIs

2010-06-12 Thread Voip Asterisk

Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi to
bring up the PRIs without alarms.

On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle supp...@drdos.info wrote:

 Voip Asterisk wrote:
  Hi,
 
  I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
   I'm using an OpenVox D410E and the drivers are loaded.  My
  system.conf looks like this:
 

 Google turned up this:

 http://www.voip-info.org/wiki/view/NFAS

 Doug

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 Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Qwest PRIs

2010-06-12 Thread Voip Asterisk

BTW these were just up and running on a Cisco AS5300, so we know they work
and Qwest has them turned up and configured correctly.  The only settings
that were needed on the cisco was the coding and framing which is how dahdi
is configured now.  Anyone have any idea how to mimic default cisco settings
on dahdi/asterisk?

The line card used in the cisco was the standard 4 port T1 PRI card.

Thanks

On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk aster...@wideideas.comwrote:

 Ya i'm not even to the asterisk part yet.  I'm still trying to get dahdi to
 bring up the PRIs without alarms.


 On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle supp...@drdos.info wrote:

 Voip Asterisk wrote:
  Hi,
 
  I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
   I'm using an OpenVox D410E and the drivers are loaded.  My
  system.conf looks like this:
 

 Google turned up this:

 http://www.voip-info.org/wiki/view/NFAS

 Doug

 --
 Ben Franklin quote:

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 Safety, deserve neither Liberty nor Safety.


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[asterisk-users] RTP ports

2010-05-03 Thread voip crazy

Hello,

I need to limit the RTP ports used by an asterisk in a client,
Actualy the range defined is from 1 to 2 udp ports.
If I only have 10 local sip extension ¿how many ports/range should I
set up in /etc/asterisk/rtp.conf?
Which is the way to calculate the rtp ports needed on an instalation?

Thanks in advance,

Voipcrazy.

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[asterisk-users] Snom Provisioning

2010-03-09 Thread voip crazy

Hello all,

I've to deploy about 200 snom320 phones on a instalation.
Do you know any knid of tool to help me with this amount of phones?
I'm thinking in a provisioning tool which I use for setting up the
phones.

Any clue would be welcomed.

Thanks.

Voip-Crazy

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[asterisk-users] ReceiveFAX G.711 + Realtime

2009-12-29 Thread Cyprus VoIP

Hello,

We're trying to receive G.711 (aLaw) faxes on the asterisk and convert 
them to tif. With T.38, we have several issues, so we are trying to use 
G.711, since the gateway is located in the same LAN, so there's no 
bandwidth/packet-lose issue.

We also use on the same Asterisk Real-Time process for the extensions.conf

My question:

Is the following syntax for disabling T.38 support correct?
vm*CLI -- Executing Set(SIP/Proxy-, t38pt_udptl=no)
vm*CLI -- Executing Set(SIP/Proxy-, SIP_CODEC=aLaw)
vm*CLI -- Executing Answer(SIP/Proxy-, )

The aLaw Set command is taken into consideration, because the SDP of the 
OK that follows these lines includes only codec 8, but when the 
ReceiveFAX command is executed, Asterisk immediately sends a T.38 reINVITE:
vm*CLI -- Executing ReceiveFAX(SIP/Proxy-, 
/var/spool/asterisk/fax/3/1261993891.0.tif)

I must be doing something wrong, but I'm not sure what :-(

Your help would be highly appreciated.

Thanks,

Andreas



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Re: [asterisk-users] ReceiveFAX G.711 + Realtime

2009-12-29 Thread Cyprus VoIP


 I have no idea where you got the idea that such a thing is possible...
 it's not. sip.conf settings for SIP endpoints are not channel variables,
 and cannot be modified from the dialplan unless the CHANNEL() dialplan
 function has been specifically extended to support them.
I was actually HOPING that it was possible, while guessing it probably 
isn't ;-), at least not like I did it.

 If you don't
 want T.38 support for a SIP endpoint, don't configure it that way in
 sip.conf or in your Realtime SIP peers table.
I want this endpoint to support T.38, but what I actually want is to 
check the initial INVITE's SDP and based on the IP address of the media, 
make a dialplan rule to decide whether to use G.711 or to switch to 
T.38. So, I found the CHANNEL() variables rtpdest and t38passthrough.

This is the dialplan real-time script I ran:
id,context,exten,priority,app,appdata
80,fax,aLaw,1,NoOp,${CHANNEL(t38passthrough)}
81,fax,aLaw,2,NoOp,${CHANNEL(rtpdest)}
82,fax,aLaw,3,Set,CHANNEL(t38passthrough)=0
83,fax,aLaw,4,GotoIf,$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]?5:T38,1
76,fax,aLaw,5,Set,SIP_CODEC=aLaw
77,fax,aLaw,6,Answer,
78,fax,aLaw,7,ReceiveFAX,${fax_filepath}/${UNIQUEID}.tif
79,fax,aLaw,8,Hangup,

The result was:
 -- Executing NoOp(SIP/Proxy-0005, 0)
 -- Executing NoOp(SIP/Proxy-0005, xx.xxx.xxx.xxx:60100)
 -- Executing Set(SIP/Proxy-0005, CHANNEL(t38passthrough)=0)
[2009-12-29 16:20:19.772] WARNING[5888]: func_channel.c:161 
func_channel_write: Unknown or unavailable item requested: 't38passthrough'
 -- Executing GotoIf(SIP/Proxy-0005, 0?5:T38,1)
 -- Goto (fax,T38,1)

Now, referring to the error above, I see (in voip-info.org) that 
t38passthrough is an R/O variable and not an R/W, but in any case, I got 
0 as a result, so it should have been OK, and it's not, as ReceiveFAX 
still sends a T.38 reINVITE. If I can't modify it, what should I do?

Also, since the rtpdest includes also the port, how do I check in the 
GotoIf if the value contains that IP and not equal to it (which it 
can't be)? It seems that this will always return 0:
$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]

Thanks.

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Re: [asterisk-users] ReceiveFAX G.711 + Realtime

2009-12-29 Thread Cyprus VoIP

 Now, referring to the error above, I see (in voip-info.org) that 
 t38passthrough is an R/O variable and not an R/W, but in any case, I got 
 0 as a result, so it should have been OK, and it's not, as ReceiveFAX 
 still sends a T.38 reINVITE. If I can't modify it, what should I do?
For the testing, I set the peer's t38pt_udptl to no and on the 
originating gateway, left only aLaw enabled. If/when I set it back to 
yes, it sent the reINVITE, so I don't have a solution for that yet.
 
 Also, since the rtpdest includes also the port, how do I check in the 
 GotoIf if the value contains that IP and not equal to it (which it 
 can't be)? It seems that this will always return 0:
 $[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]

I changed the GotoIf command to 
$[${CHANNEL(rtpdest):0:14}=xx.xxx.xxx.xxx]?5:T38,1 as I don't care about 
the length of the other originating IP addresses (the specific one I 
check is always 14 chars long), and now it stays in in aLaw context, but 
now I get this error:

app_fax.c:292 fax_generator_generate: Only generating 240 samples, where 
320 requested

I see that in app_fax.c MAX_SAMPLES is set to 240 and it doesn't seem to 
accept config file values. I reduced the packet length from 40ms (320 
samples) to 20ms (160 samples) for G.711 codecs and it solved this 
problem. Does this mean that Asterisk supports maximum 30ms packets in 
G.711 Fax?

Thanks.

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

2009-12-15 Thread Cyprus VoIP

Hello,

We upgraded the Asterisk to 1.6.1.11. Now, there's no RTP reINVITE, but 
the datagram handling of Asterisk is strange. Basically, it takes a 
commission from both ends, and ends up overflowing:

Reminder, we're dealing in this example with a passthrough, where we 
have an ATA device connected to Asterisk in the same LAN, the Asterisk 
is registered to a remote SIP Proxy server and behind it, a Fax server.

This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353

v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
---

Asterisk sends this reINVITE SDP to the ATA device (notice that the 
datagram was reduced by 2):
---
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 31812 318120001 IN IP4 192.168.2.10
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.2.10
t=0 0
m=image 4427 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:70
a=T38FaxUdpEC:t38UDPRedundancy
---

Then, it gets OK SDP from the ATA with the same settings it suggested:
---
Content-Type: application/sdp
Content-Length:   275

v=0
o=101 01 02 IN IP4 192.168.2.11
s=A conversation
c=IN IP4 192.168.2.11
t=0 0
m=image 9100 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:70
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
---

But, when it sends the OK SDP to the remote end, it lowers the datagram 
again:
---
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 936220937 936220938 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk PBX 1.6.1.11
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=image 4650 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:65
a=T38FaxUdpEC:t38UDPRedundancy
---

Then, when the ATA device sends T.38 packets, it freaks outs:
---
pbx*CLI  Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 86)
[Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL 
asked to send 77 bytes of IFP when far end only prepared to accept 12 
bytes; data loss may occur. You may need to override the 
T38FaxMaxDatagram value for this endpoint in the channel driver 
configuration.
[Dec 15 12:38:05] ERROR[5262]: udptl.c:291 encode_open_type: Buffer 
overflow detected (77 + 3  72)
[Dec 15 12:38:05] NOTICE[5262]: udptl.c:1010 ast_udptl_write: UDPTL 
Transmission error to 194.98.xxx.xxx:17548: Message too long
  Sent UDPTL packet to 194.98.xxx.xxx:17548 (type 0, seq 35, len -1)
pbx*CLI  Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 162)
[Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL 
asked to send 77 bytes of IFP when far end only prepared to accept 12 
bytes; data loss may occur. You may need to override the 
T38FaxMaxDatagram value for this endpoint in the channel driver 
configuration.
[Dec 15 12:38:05] ERROR[5262]: udptl.c:291 encode_open_type: Buffer 
overflow detected (77 + 3  72)
[Dec 15 12:38:05] NOTICE[5262]: udptl.c:1010 ast_udptl_write: UDPTL 
Transmission error to 194.98.xxx.xxx:17548: Message too long
---

Thank you for your kind assistance and support.

Regards,

Andreas

 Original Message  
Subject: Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Friday, 04 December, 2009 18:21:59

 It's probably because you are using 1.6.1.9; that release (and older)
 had a 'feature' that allowed automatic switching back to audio from T.38
 if one of the endpoints sent an audio packet. It turns out that wasn't a
 good idea, and it's been removed... but in later versions. You'll have
 to update to the latest release to get that fixed.

 
 Will do. Thanks for the explanation.

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

2009-12-15 Thread Cyprus VoIP

 Cyprus VoIP wrote:
 
 This is the reINVITE SDP received from the SIP Proxy:
 ---
 Content-Type: application/sdp
 Content-Length: 353

 v=0
 o=root 30427 30428 IN IP4 194.98.xxx.xxx
 s=session
 c=IN IP4 194.98.xxx.xxx
 t=0 0
 m=image 17548 udptl t38
 a=T38FaxVersion:0
 a=T38MaxBitRate:14400
 a=T38FaxFillBitRemoval:0
 a=T38FaxTranscodingMMR:0
 a=T38FaxTranscodingJBIG:0
 a=T38FaxRateManagement:transferredTCF
 a=T38FaxMaxBuffer:72
 a=T38FaxMaxDatagram:72
 a=T38FaxUdpEC:t38UDPRedundancy
 ---
 
 This is probably originating from a Cisco gateway. Cisco gateways
 generate T.38 SDPs that do not conform to the T.38 recommendation in one
 very obvious (and painful) way: they tell us that they can only accept
 72 byte packets (T38FaxMaxDatagram), when in fact they can accept
 packets much larger than that. When you notice that they are also
 requesting that we use t38UDPRedundancy for error correction, that means
 that the maximum IFP (single FAX protocol packet) we can include in a
 UDPTL datagram is around 30 bytes, since we'd need to have room for two
 of them and a bit of overhead. 30 bytes is a ridiculously small limit
 for IFPs, and does not allow successful FAXing at any possible bit rate
 (except for 2400 bits per second using 10 millisecond IFPs, but no FAX
 stack would do that).
 
 There is code in Asterisk already to deal with this problem, however...
 see below.
 
 pbx*CLI  Got UDPTL packet from 192.168.2.11:9100 (type 0, seq 0, len 86)
 [Dec 15 12:38:05] WARNING[5262]: udptl.c:997 ast_udptl_write: UDPTL 
 asked to send 77 bytes of IFP when far end only prepared to accept 12 
 bytes; data loss may occur. You may need to override the 
 T38FaxMaxDatagram value for this endpoint in the channel driver 
 configuration.
 
 Have you followed these instructions? The message is fairly clear in
 describing the problem, and the description of how and why this is
 needed is spelled out in the sip.conf.sample file in the configs
 directory of the source tree.
 
 Setting a lower limit for the max datagram value used when communicating
 with this peer (and others like it that generate incorrect
 T38FaxMaxDatagram values) will resolve this problem.
 

Hi,

Yes. I saw the message and the required addition in the sip.conf. The 
problem is that if I set it to 72, other terminating gateways that 
support 400 or more would also be limited to 72.

What doesn't make sense is Asterisk's commission. Why doesn't it 
simply pass/use whatever it gets without cutting the values? It looks 
like it's a bug.

Alternatively, I could (had I known how ;-)) set this ATA not to relay 
the RTP via Asterisk, in which case maybe Asterisk would leave the 
values unchanged. When I use another port on this ATA from the same 
location to the same destination without passing through the Asterisk, 
the faxes go through. How should I define this peer in sip.conf so that 
Asterisk wouldn't relay the RTP for it?

Thanks,

Andreas

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Re: [asterisk-users] ATA FXO

2009-12-14 Thread VoIP Newbie

Joseph,
You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
provides real FXO port that registers with Asterisk.
David

On Sat, Dec 12, 2009 at 1:37 AM, Joseph syscon...@gmail.com wrote:

 I'm looking for a reliable ATA FXO/FXS adapter.

 Linksys 3102 - a lot of echo problem + two of them died within a year (not
 reliable)
 Sangoma USBFXO - problem installing drive in Gentoo.

 I've tried two Chines units: AG-188N and YGW30B
 none are of them have real FXO port that will register with Asterisk.

 Any other recommendations; (I don't like internal cards).

 --
 Joseph

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[asterisk-users] Asterisk 1.6.1.11 Fax

2009-12-10 Thread Cyprus VoIP

Hello,

We're trying to receive faxes on the Asterisk server, but for the time 
being T.38 negotiation fails.

The SDP that the Asterisk reINVITE sends contains these lines:
--
m=image 4968 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy
--

The MaxDatagram and MaxBitRate are definitely not what they should be, 
and maybe other parameters are also wrong. I would like to have 400 and 
14400 respectively.

That's the udptl.conf:

--
;
; UDPTL Configuration (UDPTL is one of the transports for T.38)
;
[general]
;
; UDPTL start and UDPTL end configure start and end addresses
;
udptlstart=4000
udptlend=4999
;
; Whether to enable or disable UDP checksums on UDPTL traffic
;
;udptlchecksums=no
;
; The number of error correction entries in a UDPTL packet
;
udptlfecentries = 3
;
; The error correction type to be sent
;
;T38FaxUdpEC = t38UDPFEC
T38FaxUdpEC = t38UDPRedundancy
;
; The maximum length of a UDPTL packet
;
;T38FaxMaxBuffer = 200
T38FaxMaxDatagram = 400
VoipFaxMaxRate = 5

; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; Some VoIP providers will only accept an offer with an even-numbered
; UDPTL port. Set this option so that Asterisk will only attempt to use
; even-numbered ports when negotiating T.38. Default is no.
use_even_ports = yes
--

Can anyone help us identify the problem?

Thanks,

Andreas

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Re: [asterisk-users] Asterisk 1.6.1.11 Fax

2009-12-10 Thread Cyprus VoIP

 We're trying to receive faxes on the Asterisk server, but for the time 
 being T.38 negotiation fails.

 The SDP that the Asterisk reINVITE sends contains these lines:
 --
 m=image 4968 udptl t38
 a=T38FaxVersion:0
 a=T38MaxBitRate:9600
 a=T38FaxFillBitRemoval
 a=T38FaxTranscodingMMR
 a=T38FaxTranscodingJBIG
 a=T38FaxRateManagement:transferredTCF
 a=T38FaxMaxDatagram:1400
 a=T38FaxUdpEC:t38UDPRedundancy
 --

 The MaxDatagram and MaxBitRate are definitely not what they should be, 
 and maybe other parameters are also wrong. I would like to have 400 and 
 14400 respectively.
 
 There is no point in having a smaller T38FaxMaxDatagram value;
 Asterisk's FAX applications can handle larger packets than 400 bytes, so
  Asterisk is telling the sending endpoint that it can send them if it
 wishes to do so.
 
 As far as the T38MaxBitRate, that is only a suggestion in the
 negotiation, and rarely has any effect on the negotiation process or on
 the resulting FAX transmission.
 
 As before, you've provided only a very small amount of information, not
 enough to be able to help you determine what is wrong. Nothing in that
 SDP offer indicate any problems of any kind. In addition, you haven't
 indicated which FAX applications in Asterisk you are using (app_fax or
 res_fax), which could also have an impact on the T.38 negotiation process.
 
 If you'd like people to be able to help you debug problems, you need to
 provide enough information for them to do so; most of us are not
 clairvoyant, telepathic or omniscient. In any situation where T.38
 negotiation is failing, that means a 'sip set debug on' log trace that
 shows the entire T.38 negotiation transaction, ideally with 'core set
 debug 10' and 'core set verbose 10' as well so we can see all the
 actions that Asterisk and the FAX application took during the
 negotiation process.
 

Hi Kevin,

thank you for your response.

Before posting my question, I analyzed the entire SIP negotiation and it 
was fine. The problem began in the T.38 negotiation itself, after the 
Asterisk's reINVITE. I don't have the old calls traces anymore, but I'll 
make new ones tomorrow and post them here.

Regarding the MaxBitRate, is it possible to increase it to 14400 or is 
9600 the max allowed?

Regards,

Andreas

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[asterisk-users] Realtime Database Tables

2009-12-09 Thread Cyprus VoIP

Hello,

We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would 
like to use the sip,extensions and voicemail in realtime mode.

Where can we find the database tables structure for these versions?

Thanks,

Andreas

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Re: [asterisk-users] Realtime Database Tables

2009-12-09 Thread Cyprus VoIP

Thanks Fred,

I'm actually there, but I was wondering if the tables there are up to 
date and if any changes took place. I see all kinds of comments about 
changes.

 Original Message  
Subject: Re: [asterisk-users] Realtime Database Tables
From: Fred Posner f...@teamforrest.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Thursday, 10 December, 2009 05:26:07

 On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote:

 Hello,

 We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would 
 like to use the sip,extensions and voicemail in realtime mode.

 Where can we find the database tables structure for these versions?

 Thanks,

 Andreas

 This is the first place to go:

 http://www.voip-info.org/wiki/view/Asterisk+RealTime

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

 Cyprus VoIP wrote:
 
 Thank you for your answer. The 'internal extension' is indeed a T.38 
 capable device that works perfectly when connected directly to the 
 Proxy/ITSP.

 As you said, the key to debugging/resolving this issue is the logger. I 
 wasn't aware of this file. this is what I have there:
 ...
 ;debug = debug
 console = notice,warning,error
 ;console = notice,warning,error,debug
 messages = notice,warning,error
 ;full = notice,warning,error,debug,verbose
 ...

 Should I change the console... line or uncomment the ;full... line?
 
 Either one is fine; using 'full' is actually a bit better, because the
 color highlighting done on the console sometimes makes console captures
 hard to read.
 


Hi,

So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session

It seems that this might be the reason Asterisk initiates a reINVITE 
with voice codecs, after connecting the 2 parties.

Is there a way to disable that action, or do we need to add T.38 somehow 
to the list of codecs? I followed the instructions on the default 
sip.conf to include the line t38pt_udptl=yes,redundancy in the general 
section and in each of the parties.

Thanks.

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

 Set 'canreinvite=no' on all applicable peers?
 

I tried with yes and no. No difference. I'm almost certain it's related 
to the Keeping RTP active during T.38 session issue.

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

 Cyprus VoIP wrote:
 
 So, I enabled the full logger, and the strange thing I see is this message:
 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session

 It seems that this might be the reason Asterisk initiates a reINVITE 
 with voice codecs, after connecting the 2 parties.
 
 Sorry, that's not the issue. That just means that chan_sip didn't
 destroy the internal RTP structures used for the audio part of the call
 when the call switched to T.38, which is only an optimization so we
 don't have to recreate them if the call switches back.
 

Hi Kevin,

Thank you for your support.

If it's not related, why does Asterisk send again INVITE messages to 
both parties? How can this be prevented? I don't see more debug data 
prior to the new INVITE.

Thanks.

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

 It's probably because you are using 1.6.1.9; that release (and older)
 had a 'feature' that allowed automatic switching back to audio from T.38
 if one of the endpoints sent an audio packet. It turns out that wasn't a
 good idea, and it's been removed... but in later versions. You'll have
 to update to the latest release to get that fixed.
 

Will do. Thanks for the explanation.

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[asterisk-users] Fax throughput - Asterisk 1.6.1.9

2009-12-03 Thread Cyprus VoIP

Hello,

We are trying to send faxes by T.38 protocol to a remote SIP proxy from 
a local extension. The local extension sends the INVITE, Asterisk sends 
the call to the Proxy the call is connected with a regular audio codec. 
After a few seconds the remote proxy sends an INVITE with UDPTL and the 
Asterisk sends it to the local extension and it's accepted, but (here 
the problem starts) just after sending the OK with the proper SDP to the 
remote Proxy, the Asterisk initiates a new INVITE to the local extension 
and remote Proxy, with the normal audio codecs again.

We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the 
local extension and remote Proxy, but it still forces the call to go 
back to a voice call.

Any idea why it happens and how to debug it? We set verbose and debug to 
20, but no internal info is provided to get a clear understanding on 
Asterisk's thoughts during that process.

Thank you in advance for your assistance,

Andreas

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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

2009-12-03 Thread Cyprus VoIP

 We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the 
 local extension and remote Proxy, but it still forces the call to go 
 back to a voice call.
 
 Define 'internal extension'. Is this a T.38-capable device? If not,
 Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it is, then the
 path to resolving this problem is to collect a complete console log of
 the failing call, including 'core set debug 10', 'core set verbose 10'
 and 'sip set debug on' (and please ensure that all five logger levels
 are enabled for the 'console' log channel in logger.conf).
 

Hi Kevin,

Thank you for your answer. The 'internal extension' is indeed a T.38 
capable device that works perfectly when connected directly to the 
Proxy/ITSP.

As you said, the key to debugging/resolving this issue is the logger. I 
wasn't aware of this file. this is what I have there:
...
;debug = debug
console = notice,warning,error
;console = notice,warning,error,debug
messages = notice,warning,error
;full = notice,warning,error,debug,verbose
...

Should I change the console... line or uncomment the ;full... line?

Regards,

Andreas

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[asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

2009-11-12 Thread Cyprus VoIP

Hello,

I tried to install Asterisk + Asterisk addons + FreePBX (latest versions 
of all), but in the FreePBX screen, I don't have the option to set ring 
groups and IVRs
.
Can anyone tell me what I'm doing wrong?

Thanks,

Andreas

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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

2009-11-12 Thread Cyprus VoIP

 I tried to install Asterisk + Asterisk addons + FreePBX (latest  
 versions
 of all), but in the FreePBX screen, I don't have the option to set  
 ring
 groups and IVRs

 Can anyone tell me what I'm doing wrong?
 
 You are not posting on the FreePBX forums? ;)
 
I figured Asterisk-Users would know ;)
 
 The solution however, is to install the modules using the module admin.
 
The problem is that the online module update is not working for me 
(Cannot connect to online repository (mirror.freepbx.org). Online 
modules are not available.) and I couldn't find online a working 
solution :-(
 
 Steve
 


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Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

2009-11-12 Thread Cyprus VoIP

 The problem is that the online module update is not working for me
 (Cannot connect to online repository (mirror.freepbx.org). Online
 modules are not available.) and I couldn't find online a working
 solution :-(
 
 DNS/Gateway ok on server?
 
Yes. The problem is with the FreePBX modules. I forced the mirror file 
to include version 2.5, and I get a list, but when I try to install the 
modules, it says that the modules need FreePBX version 2.5.0alpha or rc1 
or higher, but although 2.5.2 is indeed higher, it's rejected. I've 
given up on this software and will continue to edit my .conf files 
manually. what a waste of time :-(

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Re: [asterisk-users] Music On Hold

2009-10-20 Thread Cyprus VoIP

Hello all,

I adressed this issue a couple of weekes ago, but didn't find a solution 
yet.

It seems that MOH is not initiated by Asterisk version 1.6.1.6 when 
receiving sendonly INVITE, in order to put a call on hold.

I configured features.conf with the following settings, and *9 initiates 
a proper HOLD with MOH, and so does *0 for blind transfer, but the 
a=sendonly doesn't:

[featuremap]
blindxfer = *0
atxfer = *9

Does anyone know if there's a parameter to set or if it's a bug in this 
version?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Music On Hold
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Saturday, 03 October, 2009 09:28:20

 
   What does your musiconhold.conf look like?
  
 
 
 [general]
 
 [default]
 mode=files
 directory=/var/lib/asterisk/moh
 

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[asterisk-users] Billing applications

2009-10-09 Thread voip crazy

Hello all,

I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
   - Postpaid and prepaid applications.
True CDR,

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[asterisk-users] Billing applications

2009-10-09 Thread voip crazy

Hello all,

I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
  - Postpaid and prepaid applications.
  - True CDR. Better that asterisk one, With suport for transfers
  - I do not need support for reseller
  - Billing for Voip, PSTN trunks

I need a light app. I'm not searching a heavy app. with a lots of
modules and applicacions. I need a ligth application for a soho and
its needs.

Any one are using a billing application which fits this needs?
Any clue will be welcomed.

Thanks in advance.

VoipCrazy

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Re: [asterisk-users] Music On Hold

2009-10-03 Thread Cyprus VoIP


  What does your musiconhold.conf look like?
 


[general]

[default]
mode=files
directory=/var/lib/asterisk/moh


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Re: [asterisk-users] Music On Hold

2009-10-02 Thread Cyprus VoIP

Hi,

I deleted all the default files and put one that I know that works on 
another Asterisk, but since then, I recompiled Asterisk and the default 
files were added.

In order to test moh, I created a context for it:

[default]
exten = 888,1,Goto(moh,s,1)
[moh]
exten = s,1,Answer
exten = s,2,MusicOnHold()

When we dial 888, we hear the music and this appears in the console:
 -- Executing [...@default:1] Goto(SIP/24-08650e80, moh,s,1) in 
new stack
 -- Goto (moh,s,1)
 -- Executing [...@moh:1] Answer(SIP/24-08650e80, ) in new stack
 -- Executing [...@moh:2] MusicOnHold(SIP/24-08650e80, ) in new stack
 -- Started music on hold, class 'default', on SIP/24-08650e80
 -- Stopped music on hold on SIP/24-08650e80
   == Spawn extension (moh, s, 2) exited non-zero on 'SIP/24-08650e80'


But, when I just put a call on hold, nothing is played and nothing 
appears in the console.

I have no idea why this happens and what to do about it. Any suggestions?

Thanks.


 Original Message  
Subject: Re: [asterisk-users] Music On Hold
From: John A. Sullivan III jsulli...@opensourcedevel.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Wednesday, 30 September, 2009 15:27:28

 On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
 snip
   You see the wav files but do you see the files encoded for the codecs 
 you are using?
 There's only one wav file there. No encoded files, but on asterisk 1.2 
 we have, it's the same file and it works.
 snip
 Hmm . . only one wav file.  We had several.  As I recall now, we
 actually installed 1.6.1.1 and upgraded.  1.6.1.1 had the old hold
 music.  1.6.1.6 has the new hold music.  But I believe there are several
 files.  Is that wav file valid, i.e., if you copy it to a system with a
 sound card and play it, does it play? Could it have been corrupted in
 copying or have incorrect permissions? - John

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Re: [asterisk-users] Music On Hold

2009-10-02 Thread Cyprus VoIP


 
  What is the output of moh files show CLI command ?
 


pbx*CLI moh show files
Class: default
 File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
 File: /var/lib/asterisk/moh/macroform-the_simplicity
 File: /var/lib/asterisk/moh/macroform-robot_dity
 File: /var/lib/asterisk/moh/macroform-cold_day
 File: /var/lib/asterisk/moh/reno_project-system
 File: /var/lib/asterisk/moh/music_100
 File: /var/lib/asterisk/moh/CHANGES-asterisk-moh-opsound-2
 File: /var/lib/asterisk/moh/CREDITS-asterisk-moh-opsound-2
 File: /var/lib/asterisk/moh/LICENSE-asterisk-moh-opsound-2

One more thing I tried is to add the m option in the dial command, to 
check what happens, when the call is initially originated:

exten = ,n,Dial(SIP/21SIP/22SIP/23SIP/24SIP/25,,m(default))

This is what the console shows me:

 -- Executing [1...@default:6] Dial(SIP/IN-PROXY-08563908, 
SIP/21SIP/22SIP/23SIP/24SIP/25,,m(default)) in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 21
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 22
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 23
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 24
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL CoS mark 5
 -- Called 25
 -- Started music on hold, class 'default', on SIP/IN-PROXY-08563908
 -- SIP/23-b7c2d928 is ringing
 -- SIP/24-b7a7b3a8 is ringing
 -- SIP/22-b7c35cd8 is ringing
 -- SIP/21-b7c3a508 is ringing
 -- SIP/25-b7a83350 is ringing
 -- Stopped music on hold on SIP/IN-PROXY-08563908
   == Spawn extension (default, 99935709, 6) exited non-zero on 
'SIP/IN-PROXY-08563908'


The music is played instead of the RBT, but during the conversation, 
when put on hold, I only get silence and I don't get any reference in 
the console to the fact that the call has been put on hold.

Thanks.

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[asterisk-users] RTP Delayed during RTCP

2009-10-01 Thread Cyprus VoIP

Hello,


Has anyone encountered that when Asterisk sends RTCP messages, it stops 
sending RTP packets until it gets an answer?


Can that be fixed?


Thanks.


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Re: [asterisk-users] Music On Hold

2009-09-30 Thread Cyprus VoIP

Hello,

We posted the question below yesterday, but got no answer from the 
community.

When we checked the same behavior with Asterisk 1.2, we got the Started 
music on hold, class... message on the console, but in 1.6, we get 
absolutely nothing.

I tried to unload and reload the moh module and everything seems normal, 
but Asterisk still doesn't respond in the console to the HOLD action, 
represented by the INVITE message. the call itself is being placed on 
hold and can be retrieved, but the audio file is not played and the held 
party hears only a silence.

If anyone knows how to debug/fix it, your help would be HIGHLY 
appreciated. We're really stuck.

Thank you all in advance.

 Original Message  
Subject: Music On Hold
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Tuesday, 29 September, 2009 14:31:28

 Hello,
 
 We need help in debugging Music On Hold on our Asterisk 1.6.1.6
 
  From the SIP debug, I see that an extension sends an INVITE of the call 
 to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but 
 I don't see in the console any reference to the call being placed on hold.
 
 When I typed moh show files, I see the wav files of the 
 /var/lib/asterisk/moh folder.
 
 How can I debug this?
 
 Thanks.

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Re: [asterisk-users] Music On Hold

2009-09-30 Thread Cyprus VoIP


  I'm afraid I can't be much help as I am both a newbie and it works just
  fine for me on 1.6.1.6.  Of course, mine was a fresh installation.
Thanks for your help, John. Mine is also a fresh installation, but now 
at least I know it's not a version issue.

  Is there anything in the logs to give you a clue?
There's absolutely nothing in the logs, and that's what surprises me.


  You see the wav files but do you see the files encoded for the codecs 
you are using?
There's only one wav file there. No encoded files, but on asterisk 1.2 
we have, it's the same file and it works.

Thanks.

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[asterisk-users] Music On Hold

2009-09-29 Thread Cyprus VoIP

Hello,

We need help in debugging Music On Hold on our Asterisk 1.6.1.6

 From the SIP debug, I see that an extension sends an INVITE of the call 
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but 
I don't see in the console any reference to the call being placed on hold.

When I typed moh show files, I see the wav files of the 
/var/lib/asterisk/moh folder.

How can I debug this?

Thanks.

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Re: [asterisk-users] Crystal Recording Interface

2009-08-31 Thread Cyprus VoIP

Hi,

Is there anyone there that installed successfully the CRI package and 
manages to play the calls listed in the call monitor page?

Regards.

 Original Message  
Subject: Re: [asterisk-users] Crystal Recording Interface
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Monday, 31 August, 2009 02:11:21

 I manage to see the calls in the monitor now, but unlike the example on 
 Tikal's site, I don't have the Play Call button next to each call, so 
 I can't listen to it. Could it be linked to the file name that I used 
 for storing the recorded calls?

  Original Message  
 Subject: Re: [asterisk-users] Crystal Recording Interface
 From: Danny Nicholas da...@debsinc.com
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 asterisk-users@lists.digium.com
 Date: Friday, 28 August, 2009 18:56:46

 The key as far as I can see is what is in your CDR database.  Once it is
 correct, everything will be as expected.  This might not be a fun hill to
 climb.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
 Sent: Friday, August 28, 2009 10:37 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Crystal Recording Interface

 I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are 
 installed and running and I have the CRI web interface available, too.

 Seems to me I just need the extensions.conf and maybe something to do 
 with CDRs to work out.

  Original Message  
 Subject: Re: [asterisk-users] Crystal Recording Interface
 From: Danny Nicholas da...@debsinc.com
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 asterisk-users@lists.digium.com
 Date: Friday, 28 August, 2009 18:22:49

 As far as I can see, the main requirements are these:
 1. An Apache install to install the software into
 2. PHP is active
 3. Your Asterisk uses a MYSQL or Postgres CDR.
 4. You have access to the database password/id.

 I'm guessing the documentation is pretty scarce since it's a
 one-trick-pony GPL offering. 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus 
 VoIP
 Sent: Friday, August 28, 2009 9:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Crystal Recording Interface

 Hello all,

 I download from Tikal's site the Crystal Recording Interface and 
 installed it on my Asterisk server, but there's no reference in the 
 installation instructions there regarding the necessary settings on 
 the Asterisk itself.

 Is anyone using it? Any detailed explanation on the implementation of 
 that solution anywhere?

 Thanks.

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[asterisk-users] Versions of Asterisk 1.6

2009-08-31 Thread Cyprus VoIP

Hello,

I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk.

Is there a clear table that describes the features and/or differences 
between them?

Are both stable enough?

Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw 
on voip-info.org that version 6 is not supported, but this information 
might be outdated.

Thanks.

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[asterisk-users] Need help - CDR MySQL

Hello all,

I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but 
without success.

Is there a proper online manual that describes all the steps to follow 
and debugging/monitoring information?

When I type in the CLI module show, cdr_addon_mysql.so is not listed, 
although in modules.conf, I added the line load = cdr_addon_mysql.so. 
I also tried preload, but it didn't change anything.

I also checked res_mysql.conf and cdr_mysql.conf, and the entire 
necessary data for the mysql server is there.

In cdr_manager.conf and cdr.conf, I set enabled = yes in [general].

I would appreciate any help I can get at this point, as I'm clueless as 
to what can be wrong.

Thanks.



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Re: [asterisk-users] Need help - CDR MySQL

Thanks. I found out that the module didn't load:

[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module: 
Error loading module 'cdr_addon_mysql.so': 
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared object 
file: No such file or directory

When I checked, I saw that it doesn't exist. It seems that when I 
installed the addons, I didn't realize that there were some issues to 
resolve first, as in the menuselect, I see that both app_addon_sql_mysql 
and cdr_addon_mysql have dependencies problems.

What should I do to resolve that?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59

 On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
 without success.

 Is there a proper online manual that describes all the steps to follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not listed,
 although in modules.conf, I added the line load = cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in [general].

 I would appreciate any help I can get at this point, as I'm clueless as
 to what can be wrong.
 
 Type:  'module load cdr_addon_mysql.so' and correct any errors you see.
 

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Re: [asterisk-users] Need help - CDR MySQL

I think that the missing component is mysqlclient, but when i yum 
update mysql, it does nothing.

Anyone know how to download the RPM? I'm using CentOS 5.3.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35

 You have to fix the dependency issues, which means install the stuff  
 you are missing that cdrmysql depends on so u can recompile it.

 Sent from my iPod

 On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:

 Thanks. I found out that the module didn't load:

 [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
 Error loading module 'cdr_addon_mysql.so':
 /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
 object
 file: No such file or directory

 When I checked, I saw that it doesn't exist. It seems that when I
 installed the addons, I didn't realize that there were some issues to
 resolve first, as in the menuselect, I see that both  
 app_addon_sql_mysql
 and cdr_addon_mysql have dependencies problems.

 What should I do to resolve that?

 Thanks.

  Original Message  
 Subject: Re: [asterisk-users] Need help - CDR MySQL
 From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sunday, 30 August, 2009 17:17:59

 On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
 but
 without success.

 Is there a proper online manual that describes all the steps to  
 follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not  
 listed,
 although in modules.conf, I added the line load =  
 cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in  
 [general].

 I would appreciate any help I can get at this point, as I'm  
 clueless as
 to what can be wrong.
 Type:  'module load cdr_addon_mysql.so' and correct any errors you  
 see.

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Re: [asterisk-users] Need help - CDR MySQL





OK, Installed all the dependencies and recompiled everything.

My cdr_addon_mysql.so loads now, but still nothing is logged in the
database.

I ran this on the CLI:
localhost*CLI cdr show status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
CDR registered backend: Adaptive ODBC
CDR registered backend: csv

Should there be a reference here to MySQL?

Anything else I should set?

Thanks.

 Original Message 
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: hh174 oliv...@hh174.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:58:48 

  
  
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
  
Olivier
  
Cyprus VoIP a crit:
  
I think that the missing component is mysqlclient, but when i "yum 
update mysql", it does nothing.

Anyone know how to download the RPM? I'm using CentOS 5.3.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35

  

  You have to fix the dependency issues, which means install the stuff  
you are missing that cdrmysql depends on so u can recompile it.

Sent from my iPod

On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:


  
Thanks. I found out that the module didn't load:

[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
Error loading module 'cdr_addon_mysql.so':
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
object
file: No such file or directory

When I checked, I saw that it doesn't exist. It seems that when I
installed the addons, I didn't realize that there were some issues to
resolve first, as in the menuselect, I see that both  
app_addon_sql_mysql
and cdr_addon_mysql have dependencies problems.

What should I do to resolve that?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59

  

  On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:

  
Hello all,

I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
but
without success.

Is there a proper online manual that describes all the steps to  
follow
and debugging/monitoring information?

When I type in the CLI "module show", cdr_addon_mysql.so is not  
listed,
although in modules.conf, I added the line "load =  
cdr_addon_mysql.so".
I also tried "preload", but it didn't change anything.

I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire
necessary data for the mysql server is there.

In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in  
"[general]".

I would appreciate any help I can get at this point, as I'm  
clueless as
to what can be wrong.
  
  
  Type:  'module load cdr_addon_mysql.so' and correct any errors you  
see.



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Re: [asterisk-users] Need help - CDR MySQL

I already did that and now cdr_addon_mysql.so is loaded, but I still 
don't get anything into the database. How can I debug it?

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Doug Lytle supp...@drdos.info
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 19:27:46

 Cyprus VoIP wrote:
 I think that the missing component is mysqlclient, but when i yum 
 update mysql, it does nothing.

   
 
 You need to make sure that mysql-devel is installed  and then re-compile 
 add-ons
 
 Doug
 
 

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[asterisk-users] Crystal Recording Interface

2009-08-28 Thread Cyprus VoIP

Hello all,


I download from Tikal's site the Crystal Recording Interface and 
installed it on my Asterisk server, but there's no reference in the 
installation instructions there regarding the necessary settings on the 
Asterisk itself.


Is anyone using it? Any detailed explanation on the implementation of 
that solution anywhere?


Thanks.


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Re: [asterisk-users] Crystal Recording Interface

2009-08-28 Thread Cyprus VoIP

I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are 
installed and running and I have the CRI web interface available, too.

Seems to me I just need the extensions.conf and maybe something to do 
with CDRs to work out.

 Original Message  
Subject: Re: [asterisk-users] Crystal Recording Interface
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Date: Friday, 28 August, 2009 18:22:49

 As far as I can see, the main requirements are these:
 1. An Apache install to install the software into
 2. PHP is active
 3. Your Asterisk uses a MYSQL or Postgres CDR.
 4. You have access to the database password/id.

 I'm guessing the documentation is pretty scarce since it's a
 one-trick-pony GPL offering.  

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
 Sent: Friday, August 28, 2009 9:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Crystal Recording Interface

 Hello all,

 I download from Tikal's site the Crystal Recording Interface and 
 installed it on my Asterisk server, but there's no reference in the 
 installation instructions there regarding the necessary settings on the 
 Asterisk itself.

 Is anyone using it? Any detailed explanation on the implementation of 
 that solution anywhere?

 Thanks.

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Re: [asterisk-users] onnecting two asterisk using B410p BRI cards

2009-08-17 Thread voip crazy

I just plug the junper in NT mode with no success.

VoipCrazy

2009/8/15 Paul Hales pdha...@optusnet.com.au:

 Use a standard network cable - but you have to activate the 'terminate'
 jumper on the NT end.

 - Also, the new BRI stuff in dahdi is much easier to work with than misdn.

 PaulH


 voip crazy wrote:
 Hello all,

 I'm trying to conect two asterisk servers using two B410p Digium
 cards. One card on each server. I just setting up the first BRI port
 on server A as nt_ptp and the first BRI port on server B as te_ptp.
 I use an ethernet wire to connect the first port of server A (nt_ptp)
 with the first port on server B (te_ptp) but the port light cotinues
 blinking on red on both sides once the cable was pluged. Then I use an
 isdn crossover wire with this king of schema and the lights get
 blinking red again.

 Tx+ 3 --+ +- 3
 .            X
 Rx+ 4 --+ +- 4
 .
 Tx- 5 --+ +--5
 .            X
 Rx- 6 --+ +--6

 In both servers when I do in asterisk CLI misdn shos stacks, the
 port one on each machine shows

 Server A:

 BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0


 Server B:

 BEGIN STACK_LIST:
  * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0

 Which kind of cable should I use?
 Why both in ports L1Link is failed?
 How could I solve that?

 Any clue will be welcomed.

 Thanks in advance.

 VoipCrazy.

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[asterisk-users] onnecting two asterisk using B410p BRI cards

2009-08-14 Thread voip crazy

Hello all,

I'm trying to conect two asterisk servers using two B410p Digium
cards. One card on each server. I just setting up the first BRI port
on server A as nt_ptp and the first BRI port on server B as te_ptp.
I use an ethernet wire to connect the first port of server A (nt_ptp)
with the first port on server B (te_ptp) but the port light cotinues
blinking on red on both sides once the cable was pluged. Then I use an
isdn crossover wire with this king of schema and the lights get
blinking red again.

Tx+ 3 --+ +- 3
.X
Rx+ 4 --+ +- 4
.
Tx- 5 --+ +--5
.X
Rx- 6 --+ +--6

In both servers when I do in asterisk CLI misdn shos stacks, the
port one on each machine shows

Server A:

BEGIN STACK_LIST:
 * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0


Server B:

BEGIN STACK_LIST:
 * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0

Which kind of cable should I use?
Why both in ports L1Link is failed?
How could I solve that?

Any clue will be welcomed.

Thanks in advance.

VoipCrazy.

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Re: [asterisk-users] INVITE Privacy Information

2009-07-28 Thread Cyprus VoIP

Thank you Philipp for your help.

I ran into this problem: When I change the CALLERID(num and name) to 
anonymous, they are also changed in the RPID line and not only in the From.

This is the script:
exten = _*67.,1,SIPAddHeader(Privacy: id);
exten = _*67.,2,Set(CALLERPRES()=prohib_passed_screen);
exten = _*67.,3,SIPAddHeader(P-Asserted-Identity: 
sip:${CALLERID(num)}...@10.10.10.20:5060);
exten = _*67.,4,Set(CALLERID(num)=anonymous)
exten = _*67.,5,Set(CALLERID(name)=Anonymous);
exten = _*67.,6,Dial(SIP/${EXTEN:3...@10.10.10.10)

... and this is the result:
INVITE sip:0011223...@10.10.10.10 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.20:5060;branch=z9hG4bK3031c135
Max-Forwards: 70
From: Anonymous sip:anonym...@10.10.10.20;tag=as6d95f136
To: sip:0011223...@10.10.10.10
Contact: sip:anonym...@10.10.10.20
Call-ID: 1732ae8f6581677c4a0d46360b102...@10.10.10.20
CSeq: 102 INVITE
User-Agent: Asterisk-PBX
Remote-Party-ID: Anonymous 
sip:anonym...@10.10.10.20;privacy=full;screen=yes
Date: Tue, 28 Jul 2009 09:00:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
P-Asserted-Identity: sip:123456...@10.10.10.20:5060
Privacy: id


 Original Message  
Subject: Re: [asterisk-users] INVITE Privacy Information
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Monday, 27 July, 2009 17:16:45

 Cyprus VoIP schrieb:
 I would like to use Asterisk to add/modify SIP headers in the INVITE 
 message, to include Privacy information, if the INVITE includes a *67 
 prefix (or another predefined prefix).

 That's an example of the INVITE I get:
 /INVITE sip:*6700112233...@192.168.1.100 SIP/2.0
 From: 123456789sip:*1234567...@192.168.1.100;tag=3
 To: sip:*6700112233...@192.168.1.100
 /
 These are the sip headers I need to add to the INVITE:
 /P-Asserted-Identity: sip:*1234567...@192.168.1.100:5060
 
 SIPAddHeader(P-Asserted-Identity: sip:@...);  // RFC 3325
 
 Remote-Party-ID: 
 sip:*1234567...@192.168.1.100:5060;party=calling;screen=yes;privacy=full
 
 Enable
 sendrpid=yes ; If Remote-Party-ID should be sent
 in the [general] section in sip.conf.
 SetCallerPres(prohib_passed_screen);
 
 Privacy: id/
 
 SIPAddHeader(Privacy: id);  // RFC 3325, RFC 3323
 
 And I need to change the From to 
 /Anonymoussip:anonym...@192.168.1.100/ and to remove the *67 
 prefix from the INVITE and To lines.
 
 Set(CALLERID(num)=anonymous);  // RFC 2543
 Set(CALLERID(name)=Anonymous);
 
 
 Philipp Kempgen

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Re: [asterisk-users] INVITE Privacy Information

2009-07-28 Thread Cyprus VoIP

That's exactly what I ended up doing:

SIPAddHeader(Remote-Party-ID: 
sip:${CALLERID(num)}...@\;privacy=full\;screen=yes)

Note the \ before each ; It wouldn't put anything behind the ; without them.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] INVITE Privacy Information
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Tuesday, 28 July, 2009 14:10:55

 Cyprus VoIP schrieb:
 I ran into this problem: When I change the CALLERID(num and name) to 
 anonymous, they are also changed in the RPID line and not only in the From.
 
 OK. I'd try to set sendrpid=no in sip.conf and then add a
 Remote-Party-ID header in the dialplan.
 SIPAddHeader(Remote-Party-ID: sip:@...;screen=yes;privacy=full);
 I have no idea if Asterisk will let you do that even if sendrpid
 is disabled.
 
  Original Message  
 Subject: Re: [asterisk-users] INVITE Privacy Information
 From: Philipp Kempgen philipp.kemp...@amooma.de
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Monday, 27 July, 2009 17:16:45

 Cyprus VoIP schrieb:
 
 These are the sip headers I need to add to the INVITE:
 /P-Asserted-Identity: sip:*1234567...@192.168.1.100:5060
 SIPAddHeader(P-Asserted-Identity: sip:@...);  // RFC 3325

 Remote-Party-ID: 
 sip:*1234567...@192.168.1.100:5060;party=calling;screen=yes;privacy=full
 Enable
 sendrpid=yes ; If Remote-Party-ID should be sent
 in the [general] section in sip.conf.
 SetCallerPres(prohib_passed_screen);
 
 And I need to change the From to 
 /Anonymoussip:anonym...@192.168.1.100/ and to remove the *67 
 prefix from the INVITE and To lines.
 Set(CALLERID(num)=anonymous);  // RFC 2543
 Set(CALLERID(name)=Anonymous);
 
 
 Philipp Kempgen

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[asterisk-users] INVITE Privacy Information

2009-07-27 Thread Cyprus VoIP


Hello all,

I would like to use Asterisk to add/modify SIP headers in the INVITE 
message, to include Privacy information, if the INVITE includes a *67 
prefix (or another predefined prefix).


That's an example of the INVITE I get:
/INVITE sip:*6700112233...@192.168.1.100 SIP/2.0
From: 123456789sip:*1234567...@192.168.1.100;tag=3
To: sip:*6700112233...@192.168.1.100
/
These are the sip headers I need to add to the INVITE:
/P-Asserted-Identity: sip:*1234567...@192.168.1.100:5060
Remote-Party-ID: 
sip:*1234567...@192.168.1.100:5060;party=calling;screen=yes;privacy=full

Privacy: id/

And I need to change the From to 
/Anonymoussip:anonym...@192.168.1.100/ and to remove the *67 
prefix from the INVITE and To lines.


How can I do it?

Thanks.

Thierry
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[asterisk-users] Manipulating REGISTER messages

2009-03-22 Thread Cyprus VoIP

Hello,

I would like to add SIP headers to the REGISTER messages Asterisk (1.6)
sends to an external proxy.

Also, I want to be able to reorder the lines.

Is it possible?

If yes, how?

Thanks.
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[asterisk-users] Printing faxes

2009-03-12 Thread voip crazy

Hello list,

I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.

¿Anybody have this feature implementing in their systems?

¿How is the best way to get that?

Any clue will be welcomed.

Thanks.

VoipCrazy

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[asterisk-users] Webcall app needed

2009-01-27 Thread voip crazy

Hello all,

I need to configure an application which let me to call from a web page.

Someone has experience using apps to make webcalls?
Which software do you use?

Thanks.

VoipCrazy.

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[asterisk-users] Hylafax asterisk iaxmodem problem

2008-10-23 Thread voip crazy

Hello all,

I have an asterisk box running in a customer with Hylafax, iaxmodem,
asterisk 1.2.18.

The service can receive faxes, from a lot of fax machines, but there
are a couple of them that asterisk Hylafax cannot complete.

This calls arrive the asterisk box, asterisk detect that this calls
are fax, asterisk answer the call, and then Hangup the call. But
hylafax do not receive nothing.

When I run zap show channel 1, on the asterisk CLI. The outpuit shows,


File Descriptor: 20
Span: 2
Extension:
Dialing: no
Context: from-zaptel
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Offhook

Why some faxes do not get received?
What could be wrong?

Any clue wil be welcomed.

Thanks in advanced.

VoipCrazy.

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[asterisk-users] WebCall application

2008-10-22 Thread voip crazy

Hello list,

Does anybody know any free WebCall solution to let our customer call
us directly via our web site?

Any clue will be welcomed.

Thanks.

VoipCrazy

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[asterisk-users] B410p question

2008-10-02 Thread voip crazy

Hello list,

I have got an asterisk box installed working ok with an b410p card to
make and receive isdn calls.
All works ok, but when a call is answer and the person starts to
speak, always I can ear a beep during the call. This beep is ear
some times in about 30 seconds between each beep.

Pasted bellow I send /etc/misdn-init.conf and /etc/asterisk/misdn.conf

Any clue will be apreciated.

Thanks.

VoipCrazy


- My /etc/misdn-init.conf -

#
# Configuration file for your misdn hardware
#
# Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help
#

#
# Card Settings
#
# Syntax: card=number,type[,option...]
#
#number   count your cards beginning with 1
#type either 0x1,0x4 or 0x8 for your hfcmulti hardware,
#   or the name of your card driver module.
#option   ulaw   - uLaw (instead of aLaw)
#   dtmf   - enable DTMF detection on all B-channels
#
#   pcm_slave  - set PCM bus into slave mode
#If you have a set of cards, all wired via PCM. Set
#all cards into pcm_slave mode and leave one out.
#The left card will automatically be Master.
#
#   ignore_pcm_frameclock   - this can be set in conjunction with
#   pcm_slave. If this card has a
#   PCI Bus Position before the Position
#   of the Master, then this port cannot
#   yet receive a frameclock, so it must
#   ignore the pcm frameclock.
#
#   rxclock- use clocking for pcm from ST Port
#   crystalclock - use clocking for pcm from PLL (genrated on board)
#   watchdog   - This dual E1 Board has a Watchdog for
#transparent mode
#
#
card=1,0x4

#
# Port settings
#
# Syntax: port_type=port_number[,port_number...]
#
#port_typete_ptp  - TE-Mode, PTP
#   te_ptmp - TE-Mode, PTMP
#   te_capi_ptp - TE-Mode (capi), PTP
#   te_capi_ptmp- TE-Mode (capi), PTMP
#   nt_ptp  - NT-Mode, PTP
#   nt_ptmp - NT-Mode, PTMP
#port_number  port that should be considered
#
#te_ptmp=1,2,3,4
#te_ptmp=1,2

te_ptp=1,2,3,4
#
# Port Options
#
# Syntax: option=port_number,option[,option...]
#
#option  master_clock  - use master clock for this S/T interface
#  (only once per chip, only for HFC 8/4)
#  optical   - optical (only HFC-E1)
#  los   - report LOS (only HFC-E1)
#  ais   - report AIS (only HFC-E1)
#  slip  - report SLIP (only HFC-E1)
#  nocrc4- turn off crc4 mode use double frame instead
#   (only HFC-E1)
#
# The master_clock option is essential for retrieving and transmitting
# faxes to avoid failures during transmission. It tells the driver to
# synchronize the Card with the given Port which should be a TE Port and
# connected to the PSTN in general.
#

option=1,master_clock

#option=2,ais,nocrc4
#option=3,optical,los,ais,slip


#
# General Options for your hfcmulti hardware
#
# poll=number
#
#Only one poll value must be given for all cards.
#Give the number of samples for each fifo process.
#By default 128 is used. Decrease to reduce delay, increase to
#reduce cpu load. If unsure, don't mess with it!!!
#Valid is 32, 64, 128, 256.
#
# dsp_poll=number
#   This is the poll option which is used by mISDN_dsp, this might
#   differ from the one given by poll= for the hfc based cards, since
#   they can only use multiples of 32, the dsp_poll is dependant on
#   the kernel timer setting which can be found in the CPU section
#   in the kernel config. Defaults are there either 100Hz, 250Hz
#   or 1000Hz. If your setting is either 1000 or 250 it is compatible
#   with the poll option for the hfc chips, if you have 100 it is
#   different and you need here a multiple of 80.
#   The default is to have no dsp_poll option, then the dsp itself
#   finds out which option is the best to use by itself
#
# pcm=number
#
#Give the id of the PCM bus. All PCM busses with the same ID
#are expected to be connected and have equal slots.
#Only one chip of the PCM bus must be master, the others slave.
#
# debug=number
#
#Enable debugging (see hfc_multi.h for debug options).
#
# dsp_options=number
#
#   set this to 2 and you'll have software bridging instead of
#   hardware bridging.
#
#
# dtmfthreshold=milliseconds
#
#   Here you can tune the sensitivity of the dtmf tone recognizer.
#
# timer=1|0
#
#   set this to 1 if you want

[asterisk-users] Asterisk Queue question

2008-10-02 Thread voip crazy

When the asterisk a queue reset their counters?

I 'm talking about this kind of info in asterisk console.

show queue 600
600  has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s

I just say that because I have a queue with strategy Fewest Calls
working for a couple of mouths, and a new agent has been added this
week in the queue and he is receiving all the incomings calls.

How could I solve that?

Thanks in advance.

VoipCrazy

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[asterisk-users] Sip Header Help

2008-10-01 Thread voip crazy

Dear List:

I need to make a sip phone (spa942) answer a call but the phone must
no ring. The user only has to show the callerId on the phone screen
without any sound.

How could I make that in asterisk? I tried to use Sip headers but I do
not know how must I say the phone don't ring when received, only shows
the callerID of the call.
How could I do that with sip header?
Which sip header should I send the phone to change the callerID of the call?

Do you know any other way to ghet that.
Any clue will be wellcomed.

Thanks for your answer.

VoipCrazy.

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Re: [asterisk-users] dundi and regcontext

2008-09-24 Thread technocrat voip

According to Your description this is a phone problem.

Asterisk behaves as its expected.

post your dundi.conf to dig more in to this.

regards
rama

On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos [EMAIL PROTECTED]wrote:

 hi,

 when a user register on my asterisk i can see it adding Noop for that
 extension, but after awhile i won't see it anymore:

 what are the reasons for it being removed on the dynamic context?
 one thing i found when i unregister it's removed.

 dialplan show myregcontext
 [ Context 'myregcontext' created by 'SIP' ]
   '100500' =   1. Noop(100500)   [SIP]
   '112802' =   1. Noop(112802)   [SIP]

 -= 2 extensions (2 priorities) in 1 context. =-

 [ Context 'pfingobizsip' created by 'SIP' ]

 -= 0 extensions (0 priorities) in 1 context. =-

 my prob is when it's removed dundi cant find it anymore so a user calling
 from server 1 cannot call user that is in server 2.

 i've set re-registration to very low (1 minute) to monitor if my phone
 re-register and to see if it will be added again on the regcontext.
 but i don't even see it unregistering after 1 minute i only unregistering
 when i am using x-lite and closing x-lite, i dont see x-lite re-registering
 if i just leave the softphone open. any idea?

 regards,
 ron


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Re: [asterisk-users] Dundi Help

2008-09-10 Thread technocrat voip

hi gior,

If i understand correctly your setup would be like below.

 A is the dundi serer
B is one pbx
C is one pbx

B and C dundi.conf contain the entity detials of A.

Either for C or B we can place calls to the extensions registered on the
other server.

When C extension make call to B extension call go through A and reach the B.

If you can provide the details of dundi.conf and the extension.conf it would
be very help full.
to dig into the issue.
could you able to do the dundi lookup ..?

regards



On Wed, Sep 10, 2008 at 3:36 PM, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:

 Hi tecnocrat,

 I 'm trying to setup a Dundi system like yours (one lookup server and 2
 pbx servers). I searched on internet for some docs but found a lot of
 stuff explaining only a part of the problem and no good example at all
 (there's a Richardson doc in internet which can help to start).
 I tried to do it myself so I generated the two keys (pri and pub) for
 each server with their own hostname then I copied:
 - .121 keys to the other two servers (.137 and .204)
 - .137 keys to .121
 - .204 keys to .121

 Let me know how if it works.

 Giorgio Incantalupo


 technocrat voip wrote:
  Hello All,
 
  Iam trying to achive a simple load balancing with dundi.
 
  Here i have three asterisk boxes like below.
 
 
  *.*.*.121  which is the dundi server
 
  *.*.*.137 A Peer which has the 1000 phone registerd to it
 
  *.*.*.204 B Peer which has the 200 phone registered to it.
 
  The expected behavior of  my setup is once i dial from 1000 phone it
  has to goto B peer using the .121 dundi server.
 
  Iam getting confused with the public key / private key stuff here.
 
  Iam using the astgenkey -n  command to generate them .
 
  Can any body help me by explaining , what keys i have to generate on
  each server and which keys i need to copy to which server .
 
  regards
 
 
 
  All the conf file stuff is is like below
 
  *.*.*.137
 
  iax.conf
 
  [priv]
  type=friend
  dbsecret=dundi/secret
  context=incomingdundi
 
 
  dundi.conf
 
  [mappings]
  priv =
  sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
 
 
  [00:0F:E2:76:4B:33]
  model = symmetric
  host = *.*.*.21
  inkey = dundincomingseven
  outkey = dundiseven
  include = priv
  permit = priv
  qualify = yes
  order = primary
 
 
  *.*.*.204
 
  iax.conf
 
  [priv]
  type=friend
  dbsecret=dundi/secret
  context=incomingdundi
 
  dundi.conf
 
  [mappings]
  priv =
  sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
  ;
 
  [00:0F:E2:76:4B:33]
  model = symmetric
  host = *.*.*.21
  inkey = dundiincomingfour
  outkey = dundifour
  include = priv
  permit = priv
  qualify = yes
  order = primary
 
  *.*.*.21
 
  iax.conf
  [priv]
  type=user
  context=local-custom
  disallow=all
  allow=ulaw
  allow=alaw
  allow=gsm
 
 
  dundi.conf
 
 
  [mappings]
  priv = dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}
 
  [00:11:0A:34:29:57]
  model = symmetric
  host = *.*.*.137
  inkey = dundi
  outkey = dundi
  include = priv
  permit = priv
  qualify = yes
  order = primary
 
  [00:11:0A:34:29:43]
  model = symmetric
  host = *.*.*.204
  inkey = dundi
  outkey = dundi
  include = priv
  permit = priv
  qualify = yes
  order = primary
  .
 
 
 
 
 
  
 
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 _
 Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
 FGA srl - http://www.fgasoftware.com -
 [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
 Tel: 02997663.14, Fax: 0291390172


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[asterisk-users] Dundi Help

2008-09-09 Thread technocrat voip

Hello All,

Iam trying to achive a simple load balancing with dundi.

Here i have three asterisk boxes like below.


*.*.*.121  which is the dundi server

*.*.*.137 A Peer which has the 1000 phone registerd to it

*.*.*.204 B Peer which has the 200 phone registered to it.

The expected behavior of  my setup is once i dial from 1000 phone it has to
goto B peer using the .121 dundi server.

Iam getting confused with the public key / private key stuff here.

Iam using the astgenkey -n  command to generate them .

Can any body help me by explaining , what keys i have to generate on each
server and which keys i need to copy to which server .

regards



All the conf file stuff is is like below

*.*.*.137

iax.conf

[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi


dundi.conf

[mappings]
priv = sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial


[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundincomingseven
outkey = dundiseven
include = priv
permit = priv
qualify = yes
order = primary


*.*.*.204

iax.conf

[priv]
type=friend
dbsecret=dundi/secret
context=incomingdundi

dundi.conf

[mappings]
priv = sipregistration,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
;

[00:0F:E2:76:4B:33]
model = symmetric
host = *.*.*.21
inkey = dundiincomingfour
outkey = dundifour
include = priv
permit = priv
qualify = yes
order = primary

*.*.*.21

iax.conf
[priv]
type=user
context=local-custom
disallow=all
allow=ulaw
allow=alaw
allow=gsm


dundi.conf


[mappings]
priv = dundi-priv-localcustom,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER}

[00:11:0A:34:29:57]
model = symmetric
host = *.*.*.137
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

[00:11:0A:34:29:43]
model = symmetric
host = *.*.*.204
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
.
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Re: [asterisk-users] Gateway errors

2008-09-05 Thread voip crazy

Thank you Hatem, I will try it now

Thanks

VoipCrazy

2008/9/2 hatem moiz [EMAIL PROTECTED]:
 you can do the following in sip .conf file

 register = username:[EMAIL PROTECTED]

 and after that write the configuration for the user:

 [ user ]
 username =
 host =
 qualify =
 secret =

 and so on, do this in the first of sip.conf file

 Best Regards

 On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote:

 Hatem,

 I cannot understan exactly what you told me.
 Could you try to explain that in other words. Better if you could post
 an example of this SIP trunk.

 thanks in advance.

 Voip Crazy



 2008/9/1 hatem moiz [EMAIL PROTECTED]:
  Asterisk is looking for a SIP trunk if you have recorded the usage of
  SIP
  trunks all it need is to find 1 SIP trunk,
 
  To fix your problem make a local sip trunk i mean sip trunk to 127.0.0.1
  and
  make sure that it is the first one in sip.conf file. OR you can make a
  sip
 
  trunk to ATA in the same lan and also be sure that it is the first trunk
  in
  sip.conf .
 
  On Mon, Sep 1, 2008 at 9:58 AM, Igor Hernandez [EMAIL PROTECTED] wrote:
 
  Thats strange, have you checked that you're not having issues with your
  router? Can you reach all the boxes in your lan while you are
  experiencing this downtime?
 
  voip crazy wrote:
   When I say extensions, I say extensions in the lan not in wan
  
   Thanks.
  
   VoipCrazy.
  
   2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
   Hello,
  
   By people do you mean people in the lan or external users?
  
   Regards,
  
   --
   Igor Hernandez
   Escape Communications
   http://www.escapetel.com
  
  
   voip crazy wrote:
   Hello list,
  
   I have an asterisk instalation with a bad internet connection cause
   this connection is down sometimes.
   When the connection is down and asterisk cannot get internet
   connection. All the extensions log out from the asterisk machine,
   and
   nobody can make any call.
  
   ¿Why if internet connection is down asterisk stops working
   correctly?
   ¿How could I solve that?
  
   Thansk.
  
   VoipCrazy
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[asterisk-users] Gateway errors

Hello list,

I have an asterisk instalation with a bad internet connection cause
this connection is down sometimes.
When the connection is down and asterisk cannot get internet
connection. All the extensions log out from the asterisk machine, and
nobody can make any call.

¿Why if internet connection is down asterisk stops working correctly?
¿How could I solve that?

Thansk.

VoipCrazy
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Re: [asterisk-users] Gateway errors

Igor,

From asterisk, when internet is down I can ping all extensions.
The same occurs in others instalations, when the internet is down, my
lical extensions log off from asterisk.

VoipCrazy


2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
 Thats strange, have you checked that you're not having issues with your
 router? Can you reach all the boxes in your lan while you are
 experiencing this downtime?

 voip crazy wrote:
 When I say extensions, I say extensions in the lan not in wan

 Thanks.

 VoipCrazy.

 2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
 Hello,

 By people do you mean people in the lan or external users?

 Regards,

 --
 Igor Hernandez
 Escape Communications
 http://www.escapetel.com


 voip crazy wrote:
 Hello list,

 I have an asterisk instalation with a bad internet connection cause
 this connection is down sometimes.
 When the connection is down and asterisk cannot get internet
 connection. All the extensions log out from the asterisk machine, and
 nobody can make any call.

 ¿Why if internet connection is down asterisk stops working correctly?
 ¿How could I solve that?

 Thansk.

 VoipCrazy
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Re: [asterisk-users] Gateway errors

When I say extensions, I say extensions in the lan not in wan

Thanks.

VoipCrazy.

2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
 Hello,

 By people do you mean people in the lan or external users?

 Regards,

 --
 Igor Hernandez
 Escape Communications
 http://www.escapetel.com


 voip crazy wrote:
 Hello list,

 I have an asterisk instalation with a bad internet connection cause
 this connection is down sometimes.
 When the connection is down and asterisk cannot get internet
 connection. All the extensions log out from the asterisk machine, and
 nobody can make any call.

 ¿Why if internet connection is down asterisk stops working correctly?
 ¿How could I solve that?

 Thansk.

 VoipCrazy
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Re: [asterisk-users] Gateway errors