[asterisk-users] SIP Trunk - problem to connect

[Marco Maximiliano Guglielmi]

Hello! Thnxs for reading!
I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset,
for instance (and it works!)
Connection parameters are:

Authentication Name: Número 11
Authentication password: 12345678
Username: 11
Display name: 11

Domain: hpbx.iplannetworks.net
Proxy serever address: 190.2.20.2
Proxy server port: 5060
Registrar server: 190.2.20.2
Registrar server port: 5060
Registration refresh time: 1800
Outbound proxy mode: Always
Outbound proxy: 190.2.20.2
Outbound proxy port: 5060

Can I add a trunk to my asteriskNOW having only this parameters?
I've tested some configurations, but I can find the correct one
Could you give me a clue or your point of view?
IPLAN tech support doesn't help with asterisk, only with phone config

One of the configs I tested without success:

register = username:passw...@hpbx.iplannetworks.net

[myprovidername]
host=hpbx.iplannetworks.net
outboundproxy=190.2.20.2
type=friend
fromuser=username
defaultuser=username
secret=password
context=from-trunk

Thanks a lot
Regards
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[asterisk-users] Can not calculate far_max_ifp before far_max_datagram has been set

[Marco Capetta]

Hello,

I use Asterisk 13.1 with a SIP trunk with a provider

When transmitting or receiving a fax in T.38 through the trunk with the 
provider I always get this warning:

WARNING [8204]: udptl.c: 852 calculate_far_max_ifp: UDPTL (no tag): Can not 
calculate far_max_ifp before far_max_datagram has been set

After the warning begins exchanging packets UDPTL, but all with sequence number 
0 or 256.
The fax is not properly transmitted or received.

I configured the trunk as follow:

t38_udptl = yes
t38_udptl_ec = redundancy
t38_udptl_maxdatagram = 400

How can I fix the problem?

Thank You
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[asterisk-users] Asterisk 13, PJSIP and T38 problem

[Marco Capetta]

Hello,

I need help to solve a problem that I am having using Asterisk 13, PJSIP and 
T38.

My setup is as follows:
SIP Provider -- Asterisk 13 -- Patton -- Physical Fax
I need to get the fax directly in T38 to Patton.

The provider sends me the fax in T38.
If I receive the T38 fax on Asterisk (using an hylafax device), I can properly 
receive the fax.
If I send a T38 fax with Asterisk (using an hylafax device) directly to the 
Patton, I receive it correctly on the physical fax.

Instead, if I route, through Asterisk, the t38 fax call received by the 
provider to the Patton, I can not receive anything.
From UDPTL debug I can see the flow of some udptl packages from the provider to 
Asterisk, but I see no udptl package from Asterisk to the Patton.
From PJSIP logs, it seems that the T38 is properly negotiated by both parties.
From the debug of the Patton i see many Wrong media type. Drop FAX Packet”.

How to ensure that the fax, received by asterisk from the provider with T38 
protocol, is routed to the Patton using again T38 protocol?

Below my pjsip.conf” file:


;===TRANSPORT=

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0:5060


;===TRUNK==

[trunk-provider]
type=endpoint
transport=simpletrans
context=in_provider
direct_media=no
disallow=all
allow=alaw
allow=g729
aors=trunk-provider
t38_udptl=yes
t38_udptl_ec=redundancy
t38_udptl_maxdatagram=400

[trunk-provider]
type=aor
contact=sip:X.X.X.X:5060

[trunk-provider]
type=identify
endpoint=trunk-provider
match=X.X.X.X
match=X.X.X.X



[trunk-patton]
type=aor
max_contacts=5

[trunk-patton]
type=endpoint
transport=simpletrans
context=in_patton
direct_media=no
disallow=all
allow=alaw
auth=trunk-patton
aors=trunk-patton
t38_udptl=yes
t38_udptl_ec=fec
t38_udptl_maxdatagram=400

[trunk-patton]
type=auth
auth_type=userpass
password=X
username=X

=

Thanks
Marco

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[asterisk-users] Debugging issues with setup

[Marco Carvalho]

Hello,

I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a 
previous server. I have replicated all the configurations, modules and setup 
that I know of. However, when I tested an outbound call, it didn’t work. 
Checking the asterisk message log yielded nothing. Any ideas on how I can 
isolate and trace the issue?

Thank you!
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[asterisk-users] R: Asterisk and Call Hold

[Marco Colombo]

Hi All,
I have a problem with asterisk and call hold.
In the re-invite package when I take the call to the hold, the SDP value  
a=sendrecv is present, according to the rfc3264 the sdp value a must be mark 
with sendonly.
I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same 
problem.
I've already read all the information about canreinvite and directmedia

Can anybody help me?

Thanks a lot
Marco
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Re: [asterisk-users] WSS over Asterisk

[Marco Signorini]

Hi

I tested yesterday the SIPML5 fix and I can confirm it works as expected 
with Asterisk 12 SVN-trunk-r415192 using chan_sip and no DTLS enabled.

Tested with Chrome 35.0.1916.153m.
The patch is targeted to Chrome. Firefox still be unable to handle calls 
in my setup.


In my tests I've found some asterisk exceptions when SIMPL5 is used from 
Chrome with the provided patch AND DTLS is configured for the peer in 
sip.conf AND certificates are installed in Chrome. I suppose this is 
something work in progress so I'm not worried about it.


I can also confirm the problem with wss where the SIPML5 seems not able 
to connect to the asterisk box.


Thank you and best regards,
Marco Signorini.



On 06/12/2014 03:21 AM, Steve Ng wrote:

I am using Asterisk v12.3.

As far as DTLS, I understand that applying the following Javascript 
will temporarily fix for SIPML5 to Asterisk: 
https://gist.github.com/steve-ng/14b9b88af43f92db1e46


WS works for me, its just wss which I'm stuck currently.


On Thu, Jun 12, 2014 at 4:37 AM, Miguel Molina 
mfmolina-lis...@millenium.com.co 
mailto:mfmolina-lis...@millenium.com.co wrote:


El 11/06/2014 1:52 p. m., Matthew Jordan escribió:




On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington
w...@willwh.com mailto:w...@willwh.com wrote:

Chrome 35 broke all of this you need to be using DTLS now
I believe.

I had working secure web sockets with asterisk 12.2.x and
chrome 34 and then google broke eveything :)

I have not yet got around to test out DTLS etc. with chrome 35

Just so I don't waste too much time when I go to test, does
anyone know if all that's required for DTLS on the asterisk
side is the following in sip.conf?

dtlsenable=yes
dtlsverify=yes
dtlsrekey=60
dtlscafile=/usr/local/share/ca-certificates/myCA.crt
dtlscertfile=/etc/ssl/mycert.com.pem
dtlssetup=actpass

I assume I also need TLS configs in http.conf


Signalling is independent of the media; DTLS only affects the media.

However, there are known issues with Chrome's negotiation of DTLS
and Asterisk - see
https://issues.asterisk.org/jira/browse/ASTERISK-22961


-- 
Matthew Jordan

Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org



It is broken in Chrome (firefox never had SDES) because the WebRTC
standard favoured the DTLS SRTP implementation instead of the SDES
one. The thing is that although Asterisk supports DTLS
implementation, it only supports SHA-1 hashing but both Firefox
and Chrome work with SHA-256. The patch proposed in ASTERISK-22961
is an effort to solve this issue.

Best regards

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[asterisk-users] sipML5, Ast12 and WebRTC: not acceptable here

[Marco Signorini]

Hi All.

I'm running some tests with the latest Asterisk SVN-branch-12-r410493M 
compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS 
machine (2.6.32-358.18.1.el6.i686).
As a client I'm using the sipMLP WebRTC javascript softphone running on 
Chrome 33.0.1750.146 m.


I have the softphone correctly registered on the Asterisk machine but as 
soon as I try to start a new call from the softphone, Asterisk answers 
with a 488 not acceptable here.


I'm probably missing something but I'm not able to find what and where. 
Is there someone able to point me to the right direction?

Below is my configuration. The sofpthone is registered as 1060.

Thanks in advance.
Marco Signorini.

pjsip.conf:

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/etc/asterisk/sslcert.pem
method=tlsv1

[1060]
type=endpoint
transport=transport-tls
context=from-internal
use_avpf=yes
media_encryption=sdes
disallow=all
allow=alaw
allow=ulaw
aors=1060
auth=1060

[1060]
type=auth
auth_type=userpass
password=1060
username=1060

[1060]
type=aor
max_contacts=10

[204]


http.conf:

enabled=yes
bindaddr=10.10.5.49
bindport=8088


CLI pjsip show endpoints

 Endpoint:  1060 Not in 
use0 of inf

 InAuth:  1060/1060
Aor:  1060  10
  Contact:  1060/sip:1060@10.10.5.106:54083;transport=ws;rt 
Unknown   nan

  Transport:  transport-tls tls  0  0 0.0.0.0:5061

 Endpoint:  204  Not in 
use0 of inf

 InAuth:  204/204
Aor:  2041
  Contact:  204/sip:204@10.10.5.120:5066;transport=udp 
Unknown   nan

  Transport:  transport-udp udp  0  0 0.0.0.0:5060



*CLI pjsip show transport transport-tls

Transport:  TransportId  Type  cos tos  
BindAddress

 
=

Transport:  transport-tls tls  0  0 0.0.0.0:5061

 ParameterName  : ParameterValue
 ==
 async_operations   : 1
 bind   : 0.0.0.0:5061
 ca_list_file   :
 cert_file  : /etc/asterisk/sslcert.pem
 cipher :
 cos: 0
 domain :
 external_media_address :
 external_signaling_address :
 external_signaling_port: 0
 local_net  :
 method : tlsv1
 password   :
 priv_key_file  :
 protocol   : tls
 require_client_cert: No
 tos: CS0
 verify_client  : No
 verify_server  : No


And this is the relevant SIP data exchange (with public IP hidden):

*CLI --- Received SIP request (2420 bytes) from WS:10.10.5.106:54411 ---
INVITE sip:204@10.10.5.49 SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport

From: John Doe (101)sip:1060@10.10.5.49;tag=heMv1HvlT7DeQxPxuqcq
To: sip:204@10.10.5.49
Contact: John Doe 
(101)sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language=en,fr,it

Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc
CSeq: 38718 INVITE
Content-Type: application/sdp
Content-Length: 1827
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5

v=0
o=- 365893986064703740 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28
m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 85.0.XXX.XXX
a=rtcp:37874 IN IP4 85.0.XXX.XXX
a=candidate:296123718 1 udp 2113937151 10.10.5.106 63858 typ host 
generation 0
a=candidate:296123718 2 udp 2113937151 10.10.5.106 63858 typ host 
generation 0
a=candidate:3103388307 1 udp 1845501695 85.0.XXX.XXX 37874 typ srflx 
raddr 10.10.5.106 rport 63858 generation 0
a=candidate:3103388307 2 udp 1845501695 85.0.XXX.XXX 37874 typ srflx 
raddr 10.10.5.106 rport 63858 generation 0

a=candidate:1596293558 1 tcp 1509957375 10.10.5.106 0 typ host generation 0
a=candidate:1596293558 2 tcp 1509957375 10.10.5.106 0 typ host generation 0
a=ice-ufrag:l8AWdK4ft+AnAYGl
a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd
a=ice-options:google-ice
a=fingerprint:sha-256 
89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10

a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 
inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo

a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0

Re: [asterisk-users] DAHDI and Oslec

[Marco Signorini]

Hi.
I had the same problem in the past and I've found that there was already 
an echo.ko module built in my kernel module folder.
I've renamed it and replaced with the one compiled with dahdi+oslec and 
it started working as expected.


I was on OpenSuse with kernel 2.6.27.56-0.1... it's very old so I can't 
tell you if this is something true for Debian 6.06 too.


Thanks.
Marco Signorini.



On 02/26/2013 05:38 PM, Doug Lytle wrote:

I'm hoping someone can help me here.

I've purchased replacement systems for 3 aging 1.4.x installs.  I'm 
hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec.


I'm also moving those installs from Mandriva 10.0 to Debian 6.06 
(Squeeze).


In my testing, the TE220P PCIe cards that I have, the timing was awful 
on both slots, so I compiled Kernel 3.6.9 from kernel.org.  Timing 
jumped to what I was expecting, so I moved on to recompiling dahdi 
complete for Oslec.


Browsed the Linux source directory for drivers/staging/echo and copied 
it to the proper tree in the dahdi complete directory.


Did a make distclean;make clean;make all

And everything compiled cleanly, including oslec.

But, when trying to set my E.C. to oslec, I get:

Feb 26 11:21:37 indyvoip modprobe: FATAL: Error inserting 
dahdi_echocan_oslec 
(/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): 
Unknown symbol in module, or unknown parameter (see dmesg)
Feb 26 11:21:37 indyvoip kernel: [ 1395.262334] dahdi_echocan_oslec: 
Unknown symbol oslec_create (err 0)
Feb 26 11:21:37 indyvoip kernel: [ 1395.262348] dahdi_echocan_oslec: 
Unknown symbol oslec_update (err 0)
Feb 26 11:21:37 indyvoip kernel: [ 1395.262365] dahdi_echocan_oslec: 
Unknown symbol oslec_free (err 0)
Feb 26 11:21:41 indyvoip modprobe: FATAL: Error inserting 
dahdi_echocan_oslec 
(/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): 
Unknown symbol in module, or unknown parameter (see dmesg)
Feb 26 11:21:41 indyvoip kernel: [ 1398.799227] dahdi_echocan_oslec: 
Unknown symbol oslec_create (err 0)
Feb 26 11:21:41 indyvoip kernel: [ 1398.799241] dahdi_echocan_oslec: 
Unknown symbol oslec_update (err 0)
Feb 26 11:21:41 indyvoip kernel: [ 1398.799258] dahdi_echocan_oslec: 
Unknown symbol oslec_free (err 0)


And dmesg shows:

[ 1395.262334] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0)
[ 1395.262348] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0)
[ 1395.262365] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0)
[ 1398.799227] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0)
[ 1398.799241] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0)
[ 1398.799258] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0)

My Googlng-Fu failed me, as everything was dated from 201 0 and 
earlier on this error.  I'm guessing that I compiled the kernel wrong, 
I followed these instructions to create .debs


http://www.howtoforge.com/kernel_compilation_ubuntu

Everything seemed to work well.

Coming from a Mandrake/Mandriva  background, I'm used to just:

make oldconfig
make menuconfig (Make my changes)
make all
make modules_install
make install

Any hints would be appreciated,

Doug



--
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Temporary Safety, deserve neither Liberty nor Safety.



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[asterisk-users] Call Hold problem

[Marco Colombo]

Hello everybody,
i have a problem with asterisk 1.8 and Call Hold
My problem is that Asterisk don't send re-invite when i pick up the call from 
hold.
I already insert canreinvite=no in all my sip channels, set dtmfmode=info in 
sip.conf and my Dial() command don't insert option like  t, T, h, H, w, 
W or L (with multiple arguments).
I already follow this discussion : 
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I run debug with asterisk, and i see that the re-invite are made by asterisk, 
but in the TO fields is present the local ip address and not the next hop ip.
This is the log : http://pastebin.com/ARUC0j4t
The asterisk IP : 87.248.56.101
The next hop IP : 87.248.56.100
Is it a bug? i'm already search on google, but i dont find anything.


Let me know, if you need more information.
Thanks for all
Best Regards
Marco

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[asterisk-users] R: R: R: Asterisk and History-Info

[Marco Colombo]

Ok, thanks for all

Best Regards
Marco

-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp
Inviato: mercoledì 26 settembre 2012 19:37
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] R: R: Asterisk and History-Info

Marco Colombo wrote:
 Hi,

Hola,

 On my invite trace I don't have history-info.

 Could you explain me how do I put history-info on SIP INVITE?

You can't. That specific RFC (4244) is not implemented within chan_sip.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:  
www.digium.com   www.asterisk.org

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[asterisk-users] Asterisk and History-Info

[Marco Colombo]

Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google, but I could not find anything...

Thanks for all
Best Regards

MC
http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature

http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature
[cid:27bf7ebe6e554490ac12463bf40df103]http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature


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[asterisk-users] R: Asterisk and History-Info

[Marco Colombo]

Hi,
Thanks for reply
What do you mean with Using flat or Realtime log files?
I need this line in the SIP Invite :

History-Info: 
sip:+3906330xx...@enter.it;user=phone;cause=302;privacy=history;index=1
History-Info: sip:+3906330X@enterSIP/2.0 100 Trying

how can I provide the data that you asked before?

Thanks
Best Regards

Da: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:34
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] Asterisk and History-Info

That may depend on the flavor of Asterisk you are using and whether you are 
using flat or realtime log files.

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Marco Colombo
Sent: Wednesday, September 26, 2012 10:33 AM
To: Asterisk-Users
Subject: [asterisk-users] Asterisk and History-Info

Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google, but I could not find anything...

Thanks for all
Best Regards

MC

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[asterisk-users] R: R: Asterisk and History-Info

[Marco Colombo]

Hi,

On my invite trace I don't have history-info.

Could you explain me how do I put history-info on SIP INVITE?


-- Executing [+39@trunk-squire-incoming:1] 
Dial(SIP/trunk-squire-outcoming-0045, SIP/) in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11186
Adding codec 14 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.20.1.2:5060:
INVITE sip:@10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8 SIP/2.0
Via: SIP/2.0/UDP yyy:5060;branch=z9hG4bK56839522;rport
Max-Forwards: 70
From: +39zzz sip:+39zzz@yyy;tag=as3e3ef2cf
To: sip:@10.20.1.2;uniq=73A845E0147AC676B88F6EC07EFF8
Contact: sip:+39zzz@yyy:5060
Call-ID: 4320ac5e6895a1c40e809ee973c7bed6@yyy:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.9.0-rc1
Date: Wed, 26 Sep 2012 18:37:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Remote-Party-ID: +39zzz 
sip:+39zzz@yyy;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 239

Thanks a lot!
Marco


Da: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:48
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: Asterisk and History-Info

Versions 1.8 and 11 (probably 10 as well) let you query SIP information. 1.2 
and 1.4 (1.6 also I think) do not.  If you are in a small environment, you can 
turn on SIP debug and put that in a separate log (would eat up the disk in a 
few days in most real environments).

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Marco Colombo
Sent: Wednesday, September 26, 2012 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: Asterisk and History-Info

Hi,
Thanks for reply
What do you mean with Using flat or Realtime log files?
I need this line in the SIP Invite :

History-Info: 
sip:+3906330xx...@enter.it;user=phone;cause=302;privacy=history;index=1
History-Info: sip:+3906330X@enterSIP/2.0 100 Trying

how can I provide the data that you asked before?

Thanks
Best Regards

Da: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com]
 Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:34
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] Asterisk and History-Info

That may depend on the flavor of Asterisk you are using and whether you are 
using flat or realtime log files.

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Marco Colombo
Sent: Wednesday, September 26, 2012 10:33 AM
To: Asterisk-Users
Subject: [asterisk-users] Asterisk and History-Info

Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google, but I could not find anything...

Thanks for all
Best Regards

MC

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[asterisk-users] R: SIP CANCEL, Reason

[Marco Colombo]

Hi Jordan,
Thanks for all, but i found this bug in Asterisk : 

https://issues.asterisk.org/jira/browse/ASTERISK-16465

Attached the patch to fix the problem, if the online site does not work.

Thanks for all
Best Regards


-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan
Inviato: giovedì 20 settembre 2012 13:42
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] SIP CANCEL, Reason


- Original Message - 

 From: Marco Colombo mcolo...@enter.it
 To: asterisk-users@lists.digium.com
 Sent: Wednesday, September 19, 2012 10:51:43 AM
 Subject: [asterisk-users] SIP CANCEL, Reason

 Hi All!
 i have a problem with asterisk 1.8.11.
 I must have in the SIP cancel message, the line “Reason”

 Example : Reason : SIP;cause=16;text=”Normal Call Clearing”

 I have already enable “use_q850_reason=yes”, but this not work.
 In my dialplan I have already add : exten =
 _X.,n,Hangup(${HANGUPCAUSE})

 Can anyone help me?
 I don’t know what to do

The use_q850_reason settings applies globally.  If you execute sip show 
settings, what is the value of the Q.850 Reason header?

If you enable 'sip set debug on', what is the actual CANCEL request sent to the 
UA?

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: 
http://digium.com  http://asterisk.org

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Index: chan_sip.c
===
--- chan_sip.c  (revision 280339)
+++ chan_sip.c  (working copy)
@@ -12514,8 +12514,19 @@
}

reqprep(resp, p, sipmethod, seqno, newbranch);
-   if (sipmethod == SIP_CANCEL  p-answered_elsewhere) {
-   add_header(resp, Reason, SIP;cause=200;text=\Call 
completed elsewhere\);
+   if (sipmethod == SIP_CANCEL) {
+   if (p-answered_elsewhere) {
+   if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON))
+   add_header(resp, Reason, 
Q.850;cause=200;text=\Call completed elsewhere\);
+   else
+   add_header(resp, Reason, 
SIP;cause=200;text=\Call completed elsewhere\);
+   }
+   else if (ast_test_flag(p-flags[1], SIP_PAGE2_Q850_REASON)  
p-hangupcause) {
+   char buf[50];
+
+   sprintf(buf, Q.850;cause=%i, p-hangupcause  0x7f);
+   add_header(resp, Reason, buf);
+   }
}

return send_request(p, resp, reliable, seqno ? seqno : p-ocseq);
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[asterisk-users] SIP CANCEL, Reason

[Marco Colombo]

Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line Reason

Example : Reason : SIP;cause=16;text=Normal Call Clearing

I have already enable use_q850_reason=yes, but this not work.
In my dialplan I have already add : exten = _X.,n,Hangup(${HANGUPCAUSE})

Can anyone help me?
I don't know what to do

Thanks for all
Best Regards



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[asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds

[Marco Mooijekind]

Dear all,

I have the following challenge using Asterisk 1.8, using a Digium B410P
card on BRI (The Netherlands, KPN ISDN) .

DAHDI is running, dahdi_tools indicates OK on my span and light on back of
card is green.

However, in Asterisk i get the following warnings every few seconds:

[Jan  2 20:33:17]   == Primary D-Channel on span 1 up
[Jan  2 20:33:26]   == Primary D-Channel on span 1 down
[Jan  2 20:33:26] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1:
D-channel is down!
[Jan  2 20:33:27]   == Primary D-Channel on span 1 up
[Jan  2 20:33:37]   == Primary D-Channel on span 1 down
[Jan  2 20:33:37] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1:
D-channel is down!
[Jan  2 20:33:38]   == Primary D-Channel on span 1 up
[Jan  2 20:33:47]   == Primary D-Channel on span 1 down
[Jan  2 20:33:47] WARNING[1671]: sig_pri.c:1095 pri_find_dchan: Span 1:
D-channel is down!
[Jan  2 20:33:48]   == Primary D-Channel on span 1 up

Result is I cannot dial out or in into Asterisk.

The asterisk module states the D Channel is going up and down every few
seconds. Some googling on Asterisk, BRI, and this warning indicates that
there might be an issue with Aterisk 1.8.x and euro BRI, version of LibPRI
etc.
has anybody experienced these problems on BRI? Any suggestions with regards
to these warnings are welcome!

Kind regards,

Marco Mooijekind.
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[asterisk-users] Problem installing B410P BRI card for asterisk

[Marco Mooijekind]

Dear all,

I know this is more a Digium hardware than an Asterisk issue. Already
posted a question at Digium, however also like to see whether anyone in the
Asterisk community has encountered the following situation:

I installed a Digium B410P BRI PCI card on my new asterisk server,
following the steps specified in the manual. I can see the PCI card is
available using the lspci command:

...
04:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit
Network Connection [8086:10d3]
05:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit
Network Connection [8086:10d3]
08:00.0 PCI bridge [0604]: ASPEED Technology, Inc. AST1150 PCI-to-PCI
Bridge [1a03:1150] (rev 02)
09:00.0 VGA compatible controller [0300]: ASPEED Technology, Inc. ASPEED
Graphics Family [1a03:2000] (rev 10)
0a:01.0 ISDN controller [0204]: Digium, Inc. Wildcard B410 quad-BRI card
[d161:b410] (rev 01)
...

I specified the following in my system.conf in /etc/dahdi:

loadzone = nl
defaultzone = nl
span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3

I loaded the driver using sudo modprobe wcb4xxp.
Next I ran dahdi_cfg -vv which returns:

DAHDI Tools Version - 2.5.0.2

DAHDI Version: 2.5.0.2
Echo Canceller(s): HWEC
Configuration
==

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02)
Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none)
(Slaves: 03)

3 channels to configure.

DAHDI_SPANCONFIG failed on span 1: No such device or address (6)

I'm in doubt about the DAHDI_SPANCONFIG failed on span 1: No such device
or address (6).
Next, if i execute sudo dmesg as specified by the manual it returns a huge
trace:


[  376.082907] Wrote 0x0 to register 0x1ab but got back 0x4
[  376.594754] Wrote 0x0 to register 0x1ab but got back 0x4
[  377.106605] Wrote 0x0 to register 0x1ab but got back 0x4
[  377.618423] Wrote 0x0 to register 0x1ab but got back 0x4
[  378.130266] Wrote 0x0 to register 0x1ab but got back 0x4
[  378.642088] Wrote 0x0 to register 0x1ab but got back 0x4
[ 1202.812870] show_signal_msg: 21 callbacks suppressed
[ 1202.812876] dahdi_tool[1277]: segfault at 3fc378fa0 ip 004021ac
sp 7fff131dd930 error 4 in dahdi_tool[40+3000]

And a lot of Wrote 0x0 to register 0x1ab but got back 0x4 statements.

If i run dahdi_tools it fails with a segmentation fault.

Any suggestions are appreciated!

Kind regards,

Marco Mooijekind.
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[asterisk-users] Dialing problem with Polycom phones after SIP update

[Marco Mooijekind]

Dear all,

I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8.
All worked well. After applying the new Polycom UC 4.0.1 software update to
the phones I notice the following:

When dialing an extension, either on- or off hook, the phone immediately
displays SIP URL:...
This does not allow me to enter a regular numeric extension.
The Polycom admin manual states that the phone displays the SIP URL input
message if the phone is not registered.
This is strange since i do see the phones registering themselves in the
Asterisk verbose logging.

Anyone experiencing this problem , any tips!

Thanks in advance!

Marco Mooijekind.
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Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

[Marco Mooijekind]

Hello Gord,

the line icon is solid black, which should indicate the lines are
registered.

Marco.


On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart gord...@gmail.com wrote:

 Does the phone show the line as registered? The little phone icon on the
 display should be solid for a registered line and just a outline for a
 unregistered line. Using wireshark to watch the SIP traffic is a easy way
 to ensure the REGISTER signally is complete.



 On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind 
 marco.mooijek...@gmail.com wrote:

 Dear all,

 I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8.
 All worked well. After applying the new Polycom UC 4.0.1 software update
 to the phones I notice the following:

 When dialing an extension, either on- or off hook, the phone immediately
 displays SIP URL:...
 This does not allow me to enter a regular numeric extension.
 The Polycom admin manual states that the phone displays the SIP URL input
 message if the phone is not registered.
 This is strange since i do see the phones registering themselves in the
 Asterisk verbose logging.

 Anyone experiencing this problem , any tips!

 Thanks in advance!

 Marco Mooijekind.

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Re: [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall

[Marco Mooijekind]

Maybe local channels will do the trick? They allow you to schedule delays
between subsequent devices ringing. Not sure whether they work as queue
members.. Marco.
Op 5 dec. 2011 16:35 schreef Sammy Govind govoi...@gmail.com het
volgende:

 Hi,
 I dont think that 2 Queue commands would help, also wrapup time is for an
 putting delay in an agent who just answered the call and hungup. I think
 for this purpose you may need to open up the source code for queue and put
 some delay in the relevant code.

 Regards,
 Sammy.

 On Mon, Dec 5, 2011 at 6:56 PM, Scott Gifford 
 sgiff...@suspectclass.comwrote:

 On Tue, Nov 22, 2011 at 5:34 PM, Douglas Mortensen 
 d...@impalanetworks.com wrote:

 Hello,

 ** **

 Does anyone have any idea of how I can program a 100ms delay in between
 the ringing of 2 subsequent calls in a queue configured with a ringall
 strategy?


 Does the wrapuptime queue option do what you want?

 http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf


 -Scott.


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Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

[Marco Mooijekind]

Maybe use a power supply instead of PoE, see if problem still occurs. Marco.
Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende:



 2011/11/30 Mike l...@net-wall.com

 Hi Olivier,

 ** **

 It if occurs only on the sidecar, I would imagine this is either a
 defective sidecar/Polycom phone, or a defective PoE switch not giving
 enough power. Changing PoE port would eliminate of confirm the PoE port
 being the issue, but I’m betting on a Polycom defect.

 ** **

 Make sure the PoE port is configured (if it`s a smart switch) to send
 maximum power to the port, with a sidecar I think the phone requires 12W.

 This info is very interesting.
 I wouldn't be too surprised that a PoE switch not supplying its theorical
 15W output on a long period.
 I'll try to use work around this possible cause by not using PoE.

 In any case, I'll report my findings here.

 

 ** **

 Regards,

 ** **

 Mike

 ** **

 ** **

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Wednesday, November 30, 2011 10:27 AM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Issue with Polycom SPIP650 and its sidecar***
 *

 ** **

 Hello,


 On one location, I've got from time to time (let say one a week) the
 following issue :
 the phone SoundPoint 650 works ok (can call or answer, display and sound
 are ok),
 the sidecar looses its display : entries on sidecar's LCD screen are not
 displayed anymore, or names are truncated, or BLF are not shown or updated.

 I only have one SPIP650 on this system so I can't compare with others.

 What could be the root cause of this ?

 Regards

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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Marco Signorini]

Hi.

I was following this thread. We normally use Patton SmartNode SN4112
series to interface to FXO ports. But I'm looking for something
different
for a future setup.
Xorcom USB channel banks seems something quite interesting. Is there
anyone that could/would share experiences using that? 
We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?

Thank you and regards,
Marco Signorini.

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sitehttp://www.ingegnitech.com
maili...@ingegnitech.com






Gilles wrote:

  On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
cur...@telecomabmex.com wrote:
  
  
	Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.

  
  
Thanks for the tip. It looks like the smallest option is 8 FXO ports:

www.xorcom.com/telephony-interfaces/astribank-models.html


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Re: [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom

[Marco Signorini]

Hi.

To open the door I'm suggesting you to use Arduino if you can't have a
PC near the door opener contacts.

Arduino is something useful for implementing this type of networked
embedded stuffs.
It's not so expensive and easy to use for people familiar with C, C++
and a little bit of electronics.
You can find it worldwide by e-commerce and is very well supported by a
lot of open sourced libraries.

If you stack an Arduino UNO and an Ethernet Shield SD you'll have a
small embedded solution providing you at maximum of four hw based TCP
sockets that you can use for implementing, for example, a little web
server. The digital input/output pins could be used (if properly
buffered by a transistor) to drive a relay to be placed in parallel with
the door opener pushbutton.
To have the best reliability you can use the tricks suggested on the
pachube website, where someone suggest to drive the Ethernet shield
reset pin at regular intervals.

At the asterisk level I've implemented something similar to what's
explained by David. The only difference is that, in order to open the
door, I've used the CURL application to generate a suitable HTTP get to
the IP address associated to the Ethernet shield on top of Arduino.

Thanks,
Marco Signorini

--
http://www.ethermania.com
http://www.ingegnitech.com


David - asterisk list wrote:
 Asterisk as a phone system makes perfect sense in a condo. You can get

 all the DID's you want and eliminate costs for the owners. You can offer
 standard FXO for people who don't care and IP sets for people who want
 to upgrade to feature sets.

 Your door openner is a piece of cake.
 1.  Create an option in your dialplan only in the from-access-door
 context that reads DTMF from the called station only.
 2. Use this to access an external program to turn on a serial port line
 for 10 seconds.
 3. This line drives a solid state relay (~$30) so you won't blow the
 sink current on the PC port that drives a standard door lock.

 A commercial door strike is about $100. The program to run the port is
 childs play. Here is a test prog I used for turning on a power hungry
 last printer. Change the comments and the sleep time and you're done.

   /*
* lpon Lineprinter ON
*  *** test program only **
*
*  (c) David Cook, 1994
*
*  Set signlal lines on serial port to turn on 5vdc
*  signal. Used for solid-state relay (low current
*  draw on RS232C port) to switch high voltage/high
*  current load for printer.
*
*  Part of an intelligent print spooler to only power
*  on/off high draw printer when required.
*
* Usage:   lpondevice  bits to set
*  For example, lpon /dev/cua4 4 to set bit 3 on
*  port /dev/cua4.
*  4 = 0100 or bit 3 which is DTR
*  2 = 0010 or bit 2 which is RTS
*  6 = 0110 or both DRT  RTS
*/
   #includesys/types.h
   #includesys/ioctl.h
   #includetermios.h
   #includefcntl.h
   #includeerrno.h
   #includestdlib.h
   #includeunistd.h
   #includestdio.h
   #includesignal.h

   #include lpswitch.h

   /* Main program. */
   int main(int argc, char **argv)
   {
 struct termios port_config;
 int fd;
 int set_bits = 2;

 /* Open monitor device. */
 if ((fd = open(SWDEV, O_RDWR | O_NDELAY))  0) {
   fprintf(stderr, lpswtich: %s: %d\n, SWDEV, strerror(errno));
   exit(1);}

 cfmakeraw(port_config );
 port_config.c_iflag=port_config.c_iflag|IXON;
 port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
 tcsetattr( fd, TCSANOW,port_config );
 ioctl(fd, TIOCMSET,set_bits );

 /* wait for printer to warm up */
 sleep(45);

 /* not say ready and release the printer */
 set_bits = 6;

 cfmakeraw(port_config );
 port_config.c_iflag=port_config.c_iflag|IXON;
 port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
 tcsetattr( fd, TCSANOW,port_config );
 ioctl(fd, TIOCMSET,set_bits );

 close(fd);
 }



 On 12/04/2011 8:16 AM, asterisk-users-requ...@lists.digium.com wrote:
  Message: 3
  Date: Mon, 11 Apr 2011 18:21:39 -0500
  From: Don Kellyd...@donkelly.biz
  Subject: Re: [asterisk-users] Asterisk as a Condo door opener/intercom
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  asterisk-users@lists.digium.com
  Message-ID:8E20A6A94C9548C8A0E27B502B18F200@DonPC
  Content-Type: text/plain;charset=us-ascii

  Continuing top posting...

  The same argument could be made for any commercial solution. Why use
  Asterisk when we could throw $4,000 at our problem for a commercial
  solution?

  I'd like to have a solution that would have the features you suggest
 for
  $400.

  --Don


  On Behalf Of C F
  Sent: Monday, April 11, 2011 11:43 AM

  Search the lists. Some hints:
  Viking electronics makes a door box that connects to any analog

[asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone

[Marco Lechner - FOSSGIS e.V.]

Hi I'm new to this list, so please forgive me off-topic or RTFM-questions.

I have an asterisk/elastix driven phone-environment using Polycom
SoundPoint IP 650 as extensions. When adding just one custom ringtone
(~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE
audio, ITU G.711 mu-law, mono 8000 Hz) the phone boots well. But after I
have chosen the custom ringtone as my ringtone the phone works without
problems until next reboot. I have to rename my custom ringtone for that
it is not found on boottime, change the ringtone to a default one and
reboot the phone to make it work again.

My questions:
1. Where are configurations done with the Webserver of the phone stored?
I guess must be somewhere in the tftpboot-dir on my asterisk/elastix
server. But I can't recognize any file changes (compared timestamps)
2. Where can I find out how many space is left on my phone (some
PDF-Guides from polycom say that about 160KB of custom ringtones or
about 120 items in contact directory are fine for the phone - I have 7
items in my contact directory and just one (57KB also tried 38KB) custom
ringtone
3. What else could be the problem for this behaviour?

Thank your for helping me gettng started with asterisk

Marco


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Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

[Marco Signorini]

Hi
Did you looked at Arduino + Ethernet Shield?
Is something you can program in C or C++ to receive a simple TCP and/or
HTTP packet and turn on an external relay.
From the dialplan you can run an http query through curl and/or an
external AGI command.

Best regards,
Marco Signorini.

--
Marco Signorini
http://www.ethermania.com
http://www.ingegnitech.com


Roberto Piola wrote:
 we're using a Damocles Mini
 (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of
 course, the damocles will have to drive a high-power relay.

 the damocles can be driven via snmp, so you'll have to simply call the
 snmpset unix standard utility

 On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades
 list-aster...@skycomuk.com wrote:
   
 Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only
 control 750W so you will probably need to get it to control a more
 powerfull relay as a heater is going to take a lot of current.
 It can be controlled by a virtual serial port so you just program the
 extension to make a system() call to a simple script which sends a
 string of characters to the serial port.

 That device is quite expensive. You could probably find something much
 cheaper on ebay.


 Gilles wrote:
 
 Hello

 I'm sure someone has already tried this: I use a couple of electric
 heaters to heat my office.

 I'd like to somehow connect them to Asterisk so that I could switch
 them on remotely by either calling the IVR or sending an e-mail to the
 Asterisk host, so that the room is warm when I get to the office :-)

 Any information appreciated.

 Thank you.


   
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[asterisk-users] Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6

[Marco Kühnel]

Hello

I recently heard this should be possible. Has anyone experience with this?

Thanks!
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Re: [asterisk-users] Problems with Dahdi 2.3.0.1 trying to load OSLEC

[Marco Signorini]

Hello Jose.

I've found the same problem on some servers and I solved it renaming (or
deleting) the echo.ko driver already present in the binary kernel
distribution:

In my system is something like:
/lib/modules/2.6.27.45-0.1-default/kernel/drivers/staging/echo/echo.ko

Hope this helps you.
Best regards,

Marco Signorini.

-- 
=
- http://www.ethermania.com - 
- http://www.ingegnitech.com - 




Jose P. Espinal wrote:
 Hello list,


 I'm facing a little issue with dahdi attempting to load the OSLEC echo 
 canceller into my current kernel.

 After compiling dahdi 2.3.0.1 with OSLEC support, I get the following 
 error when set 'oslec' as the echocanceller:

 DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22)

 - Similar errors are *NOT* present using other echo canncelers.
 - I tried adding the 'dahdi_echocan_oslec' line to /etc/dahdi/modules 
 and the error continues.

 I'm running Slackware Linux 13.0, Kernel 2.6.29.6-smp

 # dmesg
 ...
 dahdi_echocan_oslec: Unknown symbol oslec_create
 dahdi_echocan_oslec: Unknown symbol oslec_update
 dahdi_echocan_oslec: Unknown symbol oslec_free
 ...


 Could someone point me to some documentation about this incident?


 Regards,


   


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Re: [asterisk-users] SuSE Firewall2 - Port Forward Command

[Marco Signorini]

Brent A. Torrenga wrote:

 Does anyone know what commands in the config file for a SuSE Firewall
 will forward 5060 and RTP ranges to an Asterisk box in the internal LAN?

  

  

I think you need to play with the parameter FW_FORWARD_MASQ in the
/etc/sysconfig/SuSEfirewall2

For example:
FW_FORWARD_MASQ=0/0,192.168.10.1,udp,5060,80,192.168.2.3
lets you able to forward the udp 5060 from the IP 192.168.10.1 to
192.168.2.3

You need to add all the other RTP relevant rules.

Best regards.
Marco Signorini


-- 
=
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For network enthusiast people
- http://www.ethermania.com - 


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Re: [asterisk-users] audio glitches in conference

[marco . mouta]

It looks to me that u are having clock synchronism problems due to the fact you 
are using Virtual Machine so u don't have an ISDN card generating clock. Are u 
using what was called ztdummie as clock source? Can't precise the name of it in 
chan_dahdi but u have it.

What u report isn't new and is well known due to the fact u don't have a 
precise clock source for meetme..

You need to have chan_dahdi dummie... 

Hope it helps.
Marco Mouta
Enviada do dispositivo sem fios BlackBerry®

-Original Message-
From: Jeff Brower jbro...@signalogic.com
Date: Wed, 24 Feb 2010 18:25:07 
To: Jonathan Addlemanj...@redowl.ca
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] audio glitches in conference

Jonathan-

 I'm having a problem with conferences both meetme and app_conference,
 though I've done most of the testing with meetme.

 Essentially, I get little gaps in the audio - usually fewer than a dozen
 or so samples, though it does vary. They seem to occur at random, but I
 usually get one ever few seconds, on average. They also seem to delay
 some buffer somewhere, so that if I start recording (via eagi) after the
 conference has been established for half an hour or so, the stream
 received by the eagi script delayed by about 10 seconds.

How did you measure the gaps?  Using signal or speech analysis software to 
display the recording?  If you measure
number of samples between the gaps, does it correspond to multiples of RTP 
packet payload length (for example, for 8
kHz G711 multiples of 80 samples between gaps) ?

-Jeff

 First, the preliminaries: I'm on a debian lenny system, using the
 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was
 running xen, but I've shut down all the domU's to test if they were
 interfering, so now there's no sharing going on.

 I've been testing with a simple eagi script to grab the audio from the
 conference:
 #!/bin/sh
 cat /dev/fd/3  /tmp/audio.raw

 I've been testing it using the following dialplan extensions:
 [test]
 exten = testeagi,1,Answer
 exten = testeagi,n,Wait(3)
 exten = testeagi,n,EAGI(testeagi.sh)

 exten = testmeet,1,Answer
 exten = testmeet,n,MeetMe(testconf,1qd)

 exten = testsound,1,Answer
 exten = testsound,n,Playback(testbeep-asterisk)

 (testbeep is just 30 seconds of sine wave)

 I've been trying things like this:



 originate Local/testso...@test extension teste...@test

 The recorded audio plays back fine - no glitches.
 (an example is at http://www.vecotourism.org/audio17.wav)

 originate Local/teste...@test extension testm...@test
 originate Local/testso...@test extension testm...@test

 This does have the glitches.
 (an example is at http://www.vecotourism.org/audio18.wav)

 What could be causing this? And is there anything else I could be doing
 to debug it? Thanks.

 --
 Jon-o Addleman - http://www.redowl.ca


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Re: [asterisk-users] audio glitches in conference

[Marco Mouta]

Thanks to Tzafrir for the above mentiong documentation.

FYI

http://docs.tzafrir.org.il/dahdi-linux/README.html

A PBX system should generally have a single clock. If you are connected to a
telephony provider via a digital interface (e.g: E1, T1) you should also
typically use the provider's clock (as you get through the interface). Hence
one important job of Asterisk is to provide timing to the PBX.

DAHDI ticks once per millisecond (1000 times per second). On each tick
every active DAHDI channel reads and 8 bytes of data. Asterisk also uses
this for timing, through a DAHDI pseudo channel it opens.

However, not all PBX systems are connected to a telephony provider via a T1
or similar connection. With an analog connection you are not synced to the
other party. And some systems don't have DAHDI hardware at all. Even a
digital card may be used for other uses or is simply not connected to a
provider. DAHDI cards are also capable of providing timing from a clock on
card. Cheap x100P clone cards are sometimes used for that purpose.

If all the above fail, you can use the module dahdi_dummy to provide timing
alone without needing any DAHDI hardware. It will work with most systems and
kernels.

You can check the DAHDI timing source with dahdi_test, which is a small
utility that is included with DAHDI. It runs in cycles. In each such cycle
it tries to read 8192 bytes, and sees how long it takes. If DAHDI is not
loaded or you don't have the device files, it will fail immediately. If you
lack a timing device it will hang forever in the first cycle. Otherwise it
will just give you in each cycle the percent of how close it was. Also try
running it with the option -v for a verbose output.

To check the clock source that is built into dahdi_dummy, you can either
look at title of its span in /proc/dahdi file for a source: in the
description. Or even run:

strings dahdi.ko | grep source:

--
Marco Mouta


On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote:

 It looks to me that u are having clock synchronism problems due to the fact
 you are using Virtual Machine so u don't have an ISDN card generating clock.
 Are u using what was called ztdummie as clock source? Can't precise the name
 of it in chan_dahdi but u have it.

 What u report isn't new and is well known due to the fact u don't have a
 precise clock source for meetme..

 You need to have chan_dahdi dummie...

 Hope it helps.
 Marco Mouta
 Enviada do dispositivo sem fios BlackBerry®

 -Original Message-
 From: Jeff Brower jbro...@signalogic.com
 Date: Wed, 24 Feb 2010 18:25:07
 To: Jonathan Addlemanj...@redowl.ca
 Cc: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] audio glitches in conference

 Jonathan-

  I'm having a problem with conferences both meetme and app_conference,
  though I've done most of the testing with meetme.
 
  Essentially, I get little gaps in the audio - usually fewer than a dozen
  or so samples, though it does vary. They seem to occur at random, but I
  usually get one ever few seconds, on average. They also seem to delay
  some buffer somewhere, so that if I start recording (via eagi) after the
  conference has been established for half an hour or so, the stream
  received by the eagi script delayed by about 10 seconds.

 How did you measure the gaps?  Using signal or speech analysis software to
 display the recording?  If you measure
 number of samples between the gaps, does it correspond to multiples of RTP
 packet payload length (for example, for 8
 kHz G711 multiples of 80 samples between gaps) ?

 -Jeff

  First, the preliminaries: I'm on a debian lenny system, using the
  1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was
  running xen, but I've shut down all the domU's to test if they were
  interfering, so now there's no sharing going on.
 
  I've been testing with a simple eagi script to grab the audio from the
  conference:
  #!/bin/sh
  cat /dev/fd/3  /tmp/audio.raw
 
  I've been testing it using the following dialplan extensions:
  [test]
  exten = testeagi,1,Answer
  exten = testeagi,n,Wait(3)
  exten = testeagi,n,EAGI(testeagi.sh)
 
  exten = testmeet,1,Answer
  exten = testmeet,n,MeetMe(testconf,1qd)
 
  exten = testsound,1,Answer
  exten = testsound,n,Playback(testbeep-asterisk)
 
  (testbeep is just 30 seconds of sine wave)
 
  I've been trying things like this:
 
 
 
  originate Local/testso...@test extension teste...@test
 
  The recorded audio plays back fine - no glitches.
  (an example is at http://www.vecotourism.org/audio17.wav)
 
  originate Local/teste...@test extension testm...@test
  originate Local/testso...@test extension testm...@test
 
  This does have the glitches.
  (an example is at http://www.vecotourism.org/audio18.wav)
 
  What could be causing this? And is there anything else I could be doing
  to debug it? Thanks.
 
  --
  Jon-o Addleman - http://www.redowl.ca

Re: [asterisk-users] verifying correct loading of VPMADT032

[Marco Signorini]

In some motherboards I've found was not possible to assign different
IRQs through BIOS and other software ways. This was related to some
technical choices in that particular hardware.
In these situations, the only profitable solution was to swap the cards
between PCI connectors until a better configuration was found.

Regards,
Marco.

--
http://www.ingegnitech.com
http://www.ethermania.com



Greg Woods wrote:
 On Sat, 2010-01-02 at 20:25 +0100, F6HQZ wrote:

   
 cat /proc/interrupts
 Search the Digium cards drivers and look if several interfaces are using the 
 same IRQ number.
 If yes, you risk issues and data losses
 

 What can I do if there is a sharing going on? Looks like my TDM card is
 sharing it's IRQ with the video card, and I've been having some
 occasional problems with it.

 --Greg

   

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[asterisk-users] 1800 DID Provider - Suggestion

[Marco Cordeiro]

Hello All,
 
Do you guys suggest any 1800 DID Provider in the US ?
 
I'm having a hard time to find one.
 
Thanks,
 
Marco
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[asterisk-users] Asterisk Queue Agent

[Marco Sambo]

Hi all,
I have 2 question.
I have a call center queue with 5 agent; the following are the configuration
files:

*queue.conf*

[name_of_queue]
musicclass = default
announce = queue-name_of_queue
strategy = ringall
servicelevel = 60
context = callcenter
timeout = 60
retry = 5
wrapuptime=15
autopause=no
maxlen = 0
announce-frequency = 60
periodic-announce-frequency=30
announce-holdtime = yes
announce-round-seconds = 10
queue-youarenext=queue-youarenext  ; (You are now first in line.)
queue-thereare=queue-thereare  ; (There are)
queue-callswaiting=queue-callswaiting  ; (calls waiting.)
queue-holdtime=queue-holdtime  ; (The current est. holdtime
is)
queue-minutes=queue-minutes; (minutes.)
queue-seconds=queue-seconds; (seconds.)
queue-thankyou=queue-thankyou  ; (Thank you for your
patience.)
queue-lessthan=queue-less-than ; (less than)
queue-reporthold=queue-reporthold  ; (Hold time)
periodic-announce=queue-periodic-announce  ; (All reps busy / wait for
next)
reportholdtime = yes
ringinuse = no
memberdelay = 3
timeoutrestart = yes

monitor-format = wav
monitor-join = no

member=agent/1,,Agent 1
member=agent/2,,Agent 2
member=agent/3,,Agent 3
member=agent/4,,Agent 4
member=agent/5,,Agent 5


*agent.conf*

[general]
persistentagents=yes

[agents]
maxlogintries=3
autologoffunavail=yes
ackcall=always
endcall=no
wrapuptime=5000
musiconhold = default
updatecdr=yes

;recordagentcalls=yes
;recordformat=wav
;urlprefix=CALLCENTER
;savecallsin=/var/calls
custom_beep=beep

agent= 1,1234,Agent 1
agent= 2,1234,Agent 2
agent= 3,1234,Agent 3
agent= 4,1234,Agent 4
agent= 5,1234,Agent 5



*FIRST QUESTION*:
if I comment in agent.conf parameters recordagentcalls I can record
conversion formed by 2 file (side in and side out) with the name that I
choose by ${MONITOR_FILENAME}, but I loose the information which agent
answer.
If I uncomment in agent.conf the parameter recordagentcalls I can view on
file name which agent answer, but I can't choose postfix file name and I
can't record the two side (in  out) audio files.
Someone can help me to record two side (in  out) audio name, with agent id
and a predefined postfix file name 

*SECOND QUESTION*:
how can I set the queue to play an estimated hold time in queue to the
member in the queue  I can play only to agent. Someone can help me 




Thanks to all for your help

Marco
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[asterisk-users] lawnmower man attack sip tag=Zerogij34 some one else notice this in 20th september or recently?

[Marco Mouta]

Dear all,

According to:

http://www.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/

Citibankhas been under a telephone calling attack in 20th september.

Does anyone in asterisk community got any CDRs or logging of similar
attacks as the one above mentioned ?

Any one  with logging of it or future information about the case ?
Identified more detaills in this attack ?



Citibank is or has been under a telephone calling attack latest 12
hours. Here I will explain the attack and how it was done.


Have you seen the movie “lawnmower man”, when in the end, all phones
rings in the who city? This was the aim for todays attack on Citibank
in UK. The attack was simple, but probably effective when it was
active. Send SIP INVITE to open SIP gateways and PBXs, who then will
actually use the traditional phonesystem (POTS) to call the target.
Suddenly you need DoS protection on your traditional POTS lines….

The SIP INVITE looks like this.

INVITE sip:00442075005...@x SIP/2.0
Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg
To: sip:x
From: sip:217.23.7.47:58585;tag=Zerogij34
Call-ID: 213948958-34384780214-384...@217.23.7.47
CSeq: 1 INVITE
Max-Forwards: 69
Contact: sip:s...@217.23.7.47:58585;transport=udp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 520
Session-Expires: 3600;
Allow-Events: refer..
      v=0
      o=sip 2147483647 1 IN IP4 1.1.1.1
      s=sip
      c=IN IP4 1.1.1.1
      t=0 0
      m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98
      a=rtpmap:96 telephone-event/8000
      a=sendrecva=ptime:20
      a=rtpmap:18 G729AB/8000
      a=rtpmap:18 G729B/8000
      a=rtpmap:18 G729A/8000
      a=rtpmap:18 G729/8000
      a=rtpmap:4 G723

Lets walk through the SIP packet and see what info we can get from it:

A quick google search on the tag: Zerogij34 reveals that this attack
has been around since at least 6th of August.

The IP (217.23.7.47)from this packet should be located in Portugal but
the other attacks originate from both UK and Netherlands.
There is no User-Agent listed, so the packet is very likely crafted
from toosl like sipsak or sipp.
The codec list seems real, but they use an obscure address (1.1.1.1)
for the RTP. If they would use their own IP address, it could case a
small DoS with RTP traffic for every successful call.)The port 29784
is within the range of Cisco units (26 000-32 000)

The other INVITES reveals that the attacker is trying to figure the
extension to get a dial-tone:

   * INVITE sip:00442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:0442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:900442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:90442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:442075005...@67.170.104.216 SIP/2.0
   * and several more…

But is this a DoS attack on Citibank? I doubt it. Why call the
Citibank on a Sunday 5 a.m.? This is more likely that Citibank has
lots of lines and therefore the SIP INVITES does not generate an error
(busy or others). The attacker does not hear any ringtone, but he/she
should see the 180 Ringing / 180 Session in Progress. Then he or she
knows that he could actually get through to the PSTN on this SIP
proxy. If it would be a ringing attack, why does the attacker just
send one single SIP INVITE through each gateway that actually calls
this destination?

The machines with the attacking IP addresses should be put under
surveillance to see who connects to these. They are probably just some
bots in a larger network, but they need to relay back which gateways
actually responded successfully.

Sad to say, but I believe this is only the small beginning….



Looking forward to hearing from you guys ;)

Cheers,


--
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[asterisk-users] lawnmower man attack ??

[Marco Mouta]

Dear all,

According to:

w w w 
.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/

Citibankhas been under a telephone calling attack in 20th september.

Does anyone in asterisk community got any CDRs or logging of similar
attacks as the one above mentioned ?

Any one  with logging of it or future information about the case ?
Identified more detaills in this attack ?



Citibank is or has been under a telephone calling attack latest 12
hours. Here I will explain the attack and how it was done.


Have you seen the movie “lawnmower man”, when in the end, all phones
rings in the who city? This was the aim for todays attack on Citibank
in UK. The attack was simple, but probably effective when it was
active. Send SIP INVITE to open SIP gateways and PBXs, who then will
actually use the traditional phonesystem (POTS) to call the target.
Suddenly you need DoS protection on your traditional POTS lines….

The SIP INVITE looks like this.

INVITE sip:00442075005...@x SIP/2.0
Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg
To: sip:x
From: sip:217.23.7.47:58585;tag=Zerogij34
Call-ID: 213948958-34384780214-384...@217.23.7.47
CSeq: 1 INVITE
Max-Forwards: 69
Contact: sip:s...@217.23.7.47:58585;transport=udp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 520
Session-Expires: 3600;
Allow-Events: refer..
      v=0
      o=sip 2147483647 1 IN IP4 1.1.1.1
      s=sip
      c=IN IP4 1.1.1.1
      t=0 0
      m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98
      a=rtpmap:96 telephone-event/8000
      a=sendrecva=ptime:20
      a=rtpmap:18 G729AB/8000
      a=rtpmap:18 G729B/8000
      a=rtpmap:18 G729A/8000
      a=rtpmap:18 G729/8000
      a=rtpmap:4 G723

Lets walk through the SIP packet and see what info we can get from it:

A quick google search on the tag: Zerogij34 reveals that this attack
has been around since at least 6th of August.

The IP (217.23.7.47)from this packet should be located in Portugal but
the other attacks originate from both UK and Netherlands.
There is no User-Agent listed, so the packet is very likely crafted
from toosl like sipsak or sipp.
The codec list seems real, but they use an obscure address (1.1.1.1)
for the RTP. If they would use their own IP address, it could case a
small DoS with RTP traffic for every successful call.)The port 29784
is within the range of Cisco units (26 000-32 000)

The other INVITES reveals that the attacker is trying to figure the
extension to get a dial-tone:

   * INVITE sip:00442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:0442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:900442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:90442075005...@67.170.104.216 SIP/2.0
   * INVITE sip:442075005...@67.170.104.216 SIP/2.0
   * and several more…

But is this a DoS attack on Citibank? I doubt it. Why call the
Citibank on a Sunday 5 a.m.? This is more likely that Citibank has
lots of lines and therefore the SIP INVITES does not generate an error
(busy or others). The attacker does not hear any ringtone, but he/she
should see the 180 Ringing / 180 Session in Progress. Then he or she
knows that he could actually get through to the PSTN on this SIP
proxy. If it would be a ringing attack, why does the attacker just
send one single SIP INVITE through each gateway that actually calls
this destination?

The machines with the attacking IP addresses should be put under
surveillance to see who connects to these. They are probably just some
bots in a larger network, but they need to relay back which gateways
actually responded successfully.

Sad to say, but I believe this is only the small beginning….



Looking forward to hearing from you guys ;)

Cheers,


--
Marco Mouta

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[asterisk-users] SIP doesn't recognize hangup

[Marco Sambo]

Hi at all !
I've a problem and I don't know how to solve it.
My configuration is the following:

ISDN LINE --- PATTON (SIP) --- ASTERISK

in asterisk my sip.conf for sip patton account is the following:

[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
notifyringing=yes

[acc1]
context=fromPSTN_Ext1
type=friend
qualifiy=yes
host=dynamic
username=acc1
secret=1234
qualify=yes

Now I want to receive a call on acc1 and then redirect it again on acc1
through PSTN, in the following way:

[fromPSTN_Ext1]
exten = _X.,1,Noop(start call and redirect call through PSTN)
exten = _X.,n,Background(${SoundsPath}/message)
exten = _X.,n,WaitExten(2)
exten = i,n,Monitor(wav,${MONITORFILENAME},m)
exten = i,n,Dial(SIP/numbertoc...@acc1,10,r)

ISDN LINE --- PATTON (SIP acc1) --- ASTERISK --- PATTON (SIP acc1) ---
ISDN line

But if the external caller hang up the call ... the call to NUMBERTOCALL on
acc1 continue to ring until the called answer, but the call is out.

Someone can help me ?!?!?


Thanks to all


Marco
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Re: [asterisk-users] Asterisk 1.4.25 and attended transfer

[Marco Sambo]

Just done it ... and all works fine.
Thanks all.


Marco






2009/7/24 Administrator TOOTAI ad...@tootai.net

 Marco Sambo a écrit :
  Hi all,
  I've a problem: I update my asterisk to version 1.4.25, and the attended
  transfer doesn't work.
 
 [...]

 Marco,

 attented transfer are broken in 1.4.25, please upgrade to 1.4.26 (see
 changelog).

 --
 Daniel

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[asterisk-users] Asterisk 1.4.25 and attended transfer

[Marco Sambo]

Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.

A call B, B press *2 and voice announce to digit internal and select
internal of C.  CORRECT 
A hear music on hold and B talks with C.  CORRECT 
If B press *0, the call return to A.  CORRECT 
if B hangup, .. also the call hangup

Someone can help me??? Please!

Marco
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Re: [asterisk-users] how to use patgen and pattest for PRI card?

[Marco Signorini]

Hi,
I used the patgen and pattest some months ago to test two PRI cards in
two different servers connected together. I don't remember now the
details, but for sure I was using a PRI cross cable between the two
cards and a loopback connector when testing the single board.

I remember I had a lot of troubles and not so good results, even if the
two cards are still working perfectly under the normal condition (they
are in production since the end of march handling several 10 thousands
minutes a month).

Please, refer to
http://lists.digium.com/pipermail/asterisk-users/2009-March/227920.html
and to
http://lists.digium.com/pipermail/asterisk-dev/2009-March/037003.html

Best regards,
Marco Signorini

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com



Chris YM wrote:
 hello:
 I  wan to use the test tools-patgen and pattest for pri cards. 
 according to Tzafrir Cohen from
 http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know
 how to use that.
 do i need to connect two pri cards with two servers, or use a cable to
 connect two cards in one server?
 please give me a more details in term of cables and configurations.
 thanks!
 Chris

 

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[asterisk-users] USB phone with Asterisk under Linux

[Marco Sambo]

Hi all,
I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It
works: I can receive and make calls. But some buttons of USB phone don't
work properly. In particular, button *, #, and hangup have wrong key
mapping.
Someone have tried a USB phone 

Thamks all

Marco
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Re: [asterisk-users] Echo and static on PRI with errors

[Marco Signorini]

Tom O'Connor wrote:


 On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
 france...@fampeeters.com mailto:france...@fampeeters.com wrote:

 John F. Ervin wrote:
  What do you do if you find things sharing interrupts (IRQ 11) in my
  case with my X100P card.  I believe there is some sort of internal
  audio card in my cheap slow PC.
 
 Check the BIOS whether you can:
 Change the IRQ assignments
 Disable the extra hardware using the same IRQ

 Or otherwise try changing the slot it is in... I had very good results
 in the past swapping card around

 Good luck!


 I did a bit of investigation WRT the IRQ settings on this box. 

 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev
 a3) (prog-if 20)
 Subsystem: Hewlett-Packard Company Device 3207
 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11
 --
 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2
 MX/MX 400] (rev b2)
 Subsystem: Hewlett-Packard Company Device 3207
 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11
 --
 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721
 Gigabit Ethernet PCI Express (rev 11)
 Subsystem: Hewlett-Packard Company Device 3209
 Flags: bus master, fast devsel, latency 0, IRQ 11
 --
 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
 interface
 Subsystem: Device 79fe:0001
 Flags: bus master, medium devsel, latency 64, IRQ 11

 So basically there's 2 network cards and a USB controller sharing IRQ
 11 with the Openvox card. 

 I wasn't able to find any settings in the bios to manually configure
 IRQ assignments :(

 Could someone tell me how to set which IRQ the ISDN card picks up?

 -- 
 Tom O'Connor

 http://www.twinhelix.org
 t...@twinhelix.org mailto:t...@twinhelix.org
Hi,
Unfortunately is not always possible and it depends on how the mainboard
was realized. For what I can understand a lot of producers decide to
route only a subset of physical IRQ lines to the PCI slots (I think is
something related to cost reduction) and to share it with other onboard
peripherals.
This lets impossible to change the IRQ assignment for expansion cards.

This is not always true and sometimes swapping add-on cards solves the
problem.

We had better results with cards based on new Digium technology or with
Sangoma cards.

Best regards,
Marco Signorini.

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Re: [asterisk-users] Echo and static on PRI with errors

[Marco Signorini]

Dave Platt wrote:
 Could someone tell me how to set which IRQ the ISDN card picks up?
 

   
 It's a multi-stage process.

 Each PCI slot has four interrupt pins:  INTA through INTD.  A
 PCI card can choose to use any of these four (or even more than
 ..
 bridge architecture might be forcing interrupts from some cards
 to use a single line/IRQ.


   
Thank you for your complete description on how PCI IRQ subsystem works.
It's probably the best explanation I've found since years.

My warm compliments, you've my best appreciation.

Regards,
Marco Signorini.


Ingegni TECH S.r.l.
http://www.ingegnitech.com

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Re: [asterisk-users] Click-to-dial CTI for Windows

[Marco Sambo]

Hi,
I try Noojee Click and Outcall, and for my context they work fine. Some
times ago I tried SanpANumber, but it was bought by Digium and substitute
with ADA.


Bye

Marco


2009/6/15 Stefanov, Milen milen.stefa...@compuware.com

  Hello guys,

 Is there a decent click-to-dial CTI which works well with Asterisk?
 We have vanilla asterisk implementation and I have tried a few (ADA,
 Outcall etc) but they have poor documentation and don’t work very well.

 We are looking for an application which can allow us to dial a number from
 Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls
 with call history using a hardware deskphone.

 It seems simple - but nothing so far fits the bill.
 Can you recommend something?

 Thanks!
 Milen

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[asterisk-users] RES: RES: SIP Response 181 - Is it possible in A steri sk?

[Marco Cordeiro]

Hello Philipp and All,

My scenario is a bit different than the one I had explained before. I'm
sorry. 

Let's suppose I have someone calling one of my Asterisk clients. This
asterisk client has CFB (Call Forward Busy) activated. The forward number is
a Voice Mail System, however is not a Asterisk's Voice Mail. 
It is a third party Voice Mail System, that has a SIP Trunk with my Asterisk
Box. 

The test situation I have, demands having the same VMS Access number for
both, leaving a message to the Client Mail Box, or for the subscriber to
access its menu directory. 
This VMS platform will differ these two type of calls, by some change on the
invite message coming from the Asterisk. 

I was thinking about using SIP Response 181 (Call is being forwarded) as
an option to flag to the VMS letting it know, that it is supposed to treat
as a call that was diverted to it. 
But does any one have a suggestion, or real scenario similar to this that
could help me??

Thanks again,

Marco





-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada em: terça-feira, 2 de junho de 2009 14:22
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: SIP Response 181 - Is it possible in
Asteri sk?

Marco Cordeiro schrieb:
 So, what you are saying is that SIP trunks between 2 Asteriks might be
able
 to handle SIP Response 181 ?

Looks like it, but I didn't test it.

(Note to self: Here's the diff:
https://reviewboard.asterisk.org/r/201/diff/ )


 -Mensagem original-
 De: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp
Kempgen

 Philipp Kempgen schrieb:
 Marco Cordeiro schrieb:
 
 I have a situation that I have to notify the calling party that the call
 is
 being forwarded to another number. So far, in the tests that I made,
 calling
 from a SIP extension to another SIP extension with the forwarding
 activated,
 I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP
 Response 181 CALL_IS_BEING_FORWARDED).
 
 The forwarding of the SIP extensions is being set with AstDB. 
 
 My doubt is if, only a SIP Proxy would be able to trigger SIP Response
 181,
 or if it would be possible with an Asterisk Server. 
 
 IIRC Asterisk trunk can send and handle 181 Call is being forwarded.

Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -  http://www.amoocon.de
-- 

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[asterisk-users] SIP Response 181 - Is it possible in Asterisk?

[Marco Cordeiro]

Hello all,

 

I have being trying to replicate the following call scenario with my
Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html
http://www.tech-invite.com/Ti-sip-service-8.html  

 

I have a situation that I have to notify the calling party that the call is
being forwarded to another number. So far, in the tests that I made, calling
from a SIP extension to another SIP extension with the forwarding activated,
I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP
Response 181 CALL_IS_BEING_FORWARDED).

 

The forwarding of the SIP extensions is being set with AstDB. 

 

My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181,
or if it would be possible with an Asterisk Server. 

 

Thanks,

 

 

Marco Cordeiro

mhcorde...@gmail.com

 

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[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?

[Marco Cordeiro]

Thanks Philipp,

Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could
I find info about it?

Thanks again,

Marco



-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada em: terça-feira, 2 de junho de 2009 11:02
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk?

Marco Cordeiro schrieb:

 I have being trying to replicate the following call scenario with my
 Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html
 http://www.tech-invite.com/Ti-sip-service-8.html  
 
 I have a situation that I have to notify the calling party that the call
is
 being forwarded to another number. So far, in the tests that I made,
calling
 from a SIP extension to another SIP extension with the forwarding
activated,
 I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP
 Response 181 CALL_IS_BEING_FORWARDED).
 
 The forwarding of the SIP extensions is being set with AstDB. 
 
 My doubt is if, only a SIP Proxy would be able to trigger SIP Response
181,
 or if it would be possible with an Asterisk Server. 

IIRC Asterisk trunk can send and handle 181 Call is being forwarded.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -  http://www.amoocon.de
-- 

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[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?

[Marco Cordeiro]

Hi Philipp,

So, what you are saying is that SIP trunks between 2 Asteriks might be able
to handle SIP Response 181 ?

Marco


-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada em: terça-feira, 2 de junho de 2009 13:06
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk?

Philipp Kempgen schrieb:
 Marco Cordeiro schrieb:

 I have a situation that I have to notify the calling party that the call
is
 being forwarded to another number. So far, in the tests that I made,
calling
 from a SIP extension to another SIP extension with the forwarding
activated,
 I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP
 Response 181 CALL_IS_BEING_FORWARDED).
 
 The forwarding of the SIP extensions is being set with AstDB. 
 
 My doubt is if, only a SIP Proxy would be able to trigger SIP Response
181,
 or if it would be possible with an Asterisk Server. 
 
 IIRC Asterisk trunk can send and handle 181 Call is being forwarded.

However as a rule of thumb you could probably say that SIP B2BUAs
send 302 Moved temporarily whereas SIP proxies send 181 Call is
being forwarded.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -  http://www.amoocon.de
-- 

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Re: [asterisk-users] Play a file while transfering a call

[Marco Sambo]

Hi,
I do this by creating a directory waitingtransfer with only 1 file (the
audio message, the name isn't important, so you can change it everytime you
want) and then add new musiconhold class with specific waitingtransfer
directory. In your extensions.conf you change the musiconhold class to
waitingmessage class, and that's it!
For me works great!



2009/6/2 Julien Chavanton j...@atlastelecom.com

 Hi, I would like to play a file please wait while we transfer your call
 ... while dialing

 I could use music on hold (Dial CMD option m) but, the file can change
 very frequently and it could be problematic to edit musiconhold.conf and
 reload  everytime there is a new file available.

 Is there a suggestion on how to simply specify one file ? or create one
 directory with one file only without having to edit musiconhold.conf  ?

 or is there a different alternative ?




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Re: [asterisk-users] CAll-limit or incominglimit ?????

[Marco Sambo]

Hi,
in Asterisk 1.4 to limit the simoultaneous calls I use the following
parameters:

[general]
...
limitonpeers=yes
notifyringing=yes

[phone]
...
host=dynamic
username=phone
call-limit=2

So I can receive and make max 2 calls simoultaneous.
Fo me that's work fine.







2009/5/29 Yuri yuri.aster...@gmail.com

 Good morning


 How I use the described commands below to limit the number of simultaneous
 calls saw voip providers that they can be effected and be received in the
 trunk in the Freepbx?

 I verified the commands incominglimit and call-limit as I can use asterisk
 is version 1.4!

 It would like to restrict for I number it to four of calls that can be used
 in one trunk of a voip provider?


 thanks.

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Re: [asterisk-users] h extension and channel variables

[Marco Sambo]

I set a variable CalledID to ${EXTEN} before dial it. So in h extension I
can use ${CalledID}.





2009/5/26 Thomas Kenyon dig...@sanguinarius.co.uk

 On 5/26/2009 10:57, Thomas Kenyon wrote:
  Is there a method to fetch the ${EXTEN} of the channel that has been
  hung up when exten h is started?
 
  The nearest thing I can think of is to set another variable to the
  extension and pick that up. Would that be a reliable method though?
 
 Which is clearly a bad idea, since an intervening call would change this.

 My Best idea so far is to change the CallerID to the exten (although it
 may be desirable to keep it in tact, it's not as important in this case).

 Does anybody have any suggestions?

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[asterisk-users] SIP over VPN

[Marco Sambo]

Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my
notebook using with asterisk. But I have some problem with firewall and
port.
Someone knows which ports I should open on my firewall??? I can't connect
...

Thanks all.

Marco
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Re: [asterisk-users] SIP over VPN

[Marco Sambo]

Ok,
but if I want to open only SIP port on firewall, which ones? I have the
following situation:

computer A (softphone)  firewall  computer B (asterisk)

and I dont' want to open any ports, only SIP and voice.




2009/5/26 David Gibbons d...@videon-central.com

  Assuming you mean the firewall in front of the client, you don’t need to
 open any ports as long as the VPN client is tunneling all traffic to and
 from the Asterisk server.



 I  set NAT=yes in the config file for the extensions behind a VPN.



 -Dave



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
 *Sent:* Tuesday, May 26, 2009 11:21 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] SIP over VPN



 Hi all,
 I have a question. I have a VPN and I want to use a SIP softphone on my
 notebook using with asterisk. But I have some problem with firewall and
 port.
 Someone knows which ports I should open on my firewall??? I can't connect
 ...

 Thanks all.

 Marco

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[asterisk-users] High Volume US Traffic? Claim DIP Compensation!

[Marco [voicetermination.org]]

This could be a nice opportunity for users with a high volume of SIP traffic
terminating in the US: 
Collecting dip fees on outbound phone calls - fees that would otherwise go
to the local phone company.




With all the recent fees and surcharges, the cost of wholesale telecom and
dialer traffic keeps rising. But what many companies with a high volume of
IP based voice traffic don't realize is that they are able to share in these
dip fees. There is no need to switch routes or carriers in order to
participate.

 

There is a minimum of roughly 300k calls per month that terminate in the US
in order to participate. I would like to ask if you would be interested in
talking about this, if so what would be a good time and number to reach you?


 

p.s. this is not some kind of fishy scheme but a way to benefit and collect
from government telecom regulations that exist.

 

Warm Regards,

 

Marco Wind

dipfees.com

 

Ph: 646-736-7816

Tf:  888-780-0253

F :  (347) 626-2242

 

 

 

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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Marco Sambo]

FXO channels shuld have FXS signalling, and FXS channels shuld have FXO
signalling, so:

# FXO channels are 1,2,3
fxsks=1,2,3
# FXS channel is 4
fxoks=4






 sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
  that a attached fxs presents internally as a fxo
 
  I have a pstn line attached to the FXO and I have my pabx attached to
  2 FXS ports, which signal as fxo into asterisk (I could be wrong about
  that).

  # cat /etc/zaptel.conf
  fxsks=4
  fxoks=1,2,3

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[asterisk-users] FOP and UserEvent()

[Marco Sambo]

Hi all,
I try to install FOP. It's very nice.
In documentation I red that from my dial plan I can launch a popup window
with UserEvent() application.
I try to follow FOP documentation but I can't popup anything. My structure
is:
- server 1: Asterisk system
- server 2: FOP system
- client
On client I connect to FOP panel, but I don't see any popup.


Someone can help me to configure FOP popups and in the use of UserEvent()
application?

Thanks all

Marco
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[asterisk-users] Asterisk and HUD server

[Marco Sambo]

Hi,
someone has installed on an Asterisk box (not Trixbox) with Debian Linux,
the HUDlite Server?
Can someone help me in retrieve and install packages???

Thanks all

Marco
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Re: [asterisk-users] Asterisk and HUD server

[Marco Sambo]

Well, I see the rpm but my Asterisk box has Debian Linux, and I'm a little
afraid to use alien package to transform rpm to deb. Has HUDlite Server
source?? Like in tar.gz??



2009/4/23 David Klaverstyn d...@klaverstyn.com.au

  Hi Marco,



 Try this:
 http://yum.trixbox.org/centos/4/RPMS/hudlite-server-1.4.32-1.i386.rpm



 Regards

 David.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
 *Sent:* Thursday, 23 April 2009 7:29 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Asterisk and HUD server



 Hi,
 someone has installed on an Asterisk box (not Trixbox) with Debian Linux,
 the HUDlite Server?
 Can someone help me in retrieve and install packages???

 Thanks all

 Marco

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Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

[Marco Signorini]

Lee Howard wrote:
 Marco wrote:
   
 I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine.
 They are linked together through localhost. I've turned qualify on for the
 iax peer. Periodically I've this message:

 [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer:
 Peer 'iaxfax' is now UNREACHABLE! Time: 3
 [Apr 20 23:47:56] NOTICE[4632]: chan_iax2.c:8128 socket_process: Peer
 'iaxfax' is now REACHABLE! Time: 3

 It happens a lot of times during the day, even when the box is not loaded
 at all.
 

 What does iaxmodem say?  (Look at the iaxmodem logs.)
   
I've some Registration timed out events but I think they are not
related to this problem (even because they are less than the
unreachable/reachable events).
The strange think is that, when UNREACHABLE, the reported Time is 3 (I
think milliseconds) and it's the same that's reported when the peer
became reachable. Is this a little bit strange?


Thank you and best regards,
Marco Signorini.


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Re: [asterisk-users] Asterisk process ended

[Marco Sambo]

Hi,
I have the same problem: sometimes my Asterisk box crash (or similar) and in
asterisk log doesn't appear nothing. Also into syslog.
I don't understand what is it


Marco




2009/4/21 Adrien Lemoine alemo...@legos.fr

  Hi all,



 I experienced for a second time the crash of asterisk process during the
 night.



 Nothing in Asterisk messages logs, nothing in /var/log/messages can explain
 that.



 Maybe someone experienced something similar and can drive me in the
 resolution ?



 Regards,



 A.L

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[asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

[Marco]

Hi All,
I'm having a strange problem and I'm not able to understand what's happening.
I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine.
They are linked together through localhost. I've turned qualify on for the
iax peer. Periodically I've this message:

[Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer:
Peer 'iaxfax' is now UNREACHABLE! Time: 3
[Apr 20 23:47:56] NOTICE[4632]: chan_iax2.c:8128 socket_process: Peer
'iaxfax' is now REACHABLE! Time: 3

It happens a lot of times during the day, even when the box is not loaded
at all.
I've tried to connect the asterisk box to a IAXModem running on an
external PC on the same subnet and the problem was not replicable. Seems
only when the two pieces of software are running on the same machine.

Is someone having the same problem? Is there something I can do to better
understand what's the cause of this?

Thank you and best regards,
Marco Signorini.



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[asterisk-users] Remote BLF / hint on IAX2 trunk

[Marco Sambo]

Hi all,
I have a question: how can I see hints of a remote Asterisk in IAX2 trunk??
I want to set BLF on my phones to look state of other phones also in other
Asterisk server.

Someone have any idea or solution?
I use Asterisk 1.4.24.

Thanks all

Marco
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Re: [asterisk-users] Remote BLF / hint on IAX2 trunk

[Marco Sambo]

Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use
SIP trunks.
Can you help me?





2009/4/16 Philipp Kempgen philipp.kemp...@amooma.de

 Marco Sambo schrieb:
  I have a question: how can I see hints of a remote Asterisk in IAX2
 trunk??
  I want to set BLF on my phones to look state of other phones also in
 other
  Asterisk server.
 
  Someone have any idea or solution?
  I use Asterisk 1.4.24.

 Use SIP instead?


Philipp Kempgen
 --
 AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
 AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 --

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Re: [asterisk-users] Remote BLF / hint on IAX2 trunk

[Marco Sambo]

So thanks, but in Asterisk 1.4.24 is not present in any way??
Any mystique solution??

Marco



2009/4/16 Tilghman Lesher tilgh...@mail.jeffandtilghman.com

 On Thursday 16 April 2009 07:08:49 Marco Sambo wrote:
  Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use
  SIP trunks.
  Can you help me?

 Distributed device states, which is what you're talking about, will be
 available, starting in version 1.6.1.  Please see
 doc/distributed_devstate.txt
 in that tree for more information.

 --
 Tilghman

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Re: [asterisk-users] TDM2400P dial tone is not present on phones, but the phone ring with incoming calls

[Marco Sambo]

Hi, excuse me, but I see in your code that you configure DAHDI with OSLEC.
How do you do? Which version you have installed?

Thank you.

Marco




2009/4/16 Giovanni Magallanes gmagalla...@gmail.com

  Hi,

 I have a problem with TDM2400P card. The card is detected ok, I can make a
 call but only with pulse dialing (not tone dialing) without hear sounds from
 the headset. When I receive a call, I can to establish a communication, but
 without hear sounds from the headset. When I dial any phone key, I can hear
 dtmf tone.

 I'm using Elastix 1.5.2. These are my configuration files:

 http://pastebin.com/f46e05257

 Thanks,

 GM

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[asterisk-users] Asterisk and Voice Recognition Sphinx

[Marco Sambo]

Hi all,
someone has used the voice recognition software named Sphinx??? Can he tell
me how to use and its performance???

Thanks

Marco
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Re: [asterisk-users] Logging Asterisk console

[Marco Sambo]

Hi Enrico,
I do that by modifying logger.conf

[logfiles]
logpro = notice,warning,error,debug,verbose

and modifying asterisk.conf

[directories]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir = /var/lib/asterisk
astdatadir = /var/lib/asterisk
astagidir = /var/lib/asterisk/agi-bin
astspooldir = /var/spool/asterisk
astrundir = /var/run/asterisk
astlogdir = /var/log/asterisk

[options]
verbose = 3

and so I find into /var/log/asterisk the logpro file with the output of CLI
(verbose) and notice, warning, error, debug message of Asterisk.


Ciao
Marco


2009/4/7 Enrico Pasqualotto enr...@pasqualotto.org

 Hi all, in witch way can I put in a log file the asterisk console?
 I have tried with some settings in file logger.conf but the log not
 contain the same debug that I can see with asterisk -rvvv.
 I need it in debugging purpose for tracking some bug.

 Thanks Enrico.

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Re: [asterisk-users] DAHDI with OSLEC

[Marco Sambo]

But I don't have also echo

modinfo echo
modinfo: could not find module echo





2009/4/1 Dave Fullerton dfullertaster...@shorelinecontainer.com

 Marco Sambo wrote:
  Mhmm. Thaht's strange!
 
  modinfo oslec
  --
  modinfo: could not find module oslec
 
  and
 
  modinfo dahdi_echocan_oslec
  --
  filename:   /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
  license:GPL
  author: Tzafrir Cohen tzafrir.co...@xorcom.com
  description:DAHDI OSLEC wrapper
  depends:dahdi
  vermagic:   2.6.26-1-486 mod_unload modversions 486
 
 
 
 
 
 
  2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com
 
  On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote:
  Hi,
  I've a problem: I can't configure DAHDI with ech canceller OSLEC.
  I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
  But when in /etc/dahdi/systems.conf I insert value
  echocanceller=oslec,1-4,
  command dahdi_cfg - give me an error about oslec.
  What is the output of:
 
   modinfo oslec
   modinfo dahdi_echocan_oslec
 
  --
Tzafrir Cohen
  icq#16849755  
  jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 jabber%3atzafrir.co...@xorcom.com jabber%253atzafrir.co...@xorcom.com
  +972-50-7952406   mailto:tzafrir.co...@xorcom.com
  http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
  ___

 I'm not sure that is strange. When I build DAHDI with OSLEC I don't get
 an oslec module, I get an echo module:

 r...@srvpbx:~# modinfo echo
 filename:   /lib/modules/2.6.27.19-smp/staging/echo/echo.ko
 version:0.3.0
 description:Open Source Line Echo Canceller
 author: David Rowe
 license:GPL
 srcversion: 285EC80D84DCE294A677160
 depends:
 vermagic:   2.6.27.19-smp SMP preempt mod_unload 686

 r...@srvpbx:~# modinfo dahdi_echocan_oslec
 filename:   /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko
 license:GPL
 author: Tzafrir Cohen tzafrir.co...@xorcom.com
 description:DAHDI OSLEC wrapper
 depends:dahdi,echo
 vermagic:   2.6.27.19-smp SMP preempt mod_unload 686

 Try building DAHDI with the steps detailed here and see if you have
 better luck:

 http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html

 -Dave

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Re: [asterisk-users] DAHDI with OSLEC

[Marco Sambo]

One thing!
I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel
2.6.28 or newer to use oslec with DAHDI???





2009/4/1 Marco Sambo derwid...@gmail.com

 But I don't have also echo

 modinfo echo
 modinfo: could not find module echo





 2009/4/1 Dave Fullerton dfullertaster...@shorelinecontainer.com

 Marco Sambo wrote:
  Mhmm. Thaht's strange!
 
  modinfo oslec
  --
  modinfo: could not find module oslec
 
  and
 
  modinfo dahdi_echocan_oslec
  --
  filename:   /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
  license:GPL
  author: Tzafrir Cohen tzafrir.co...@xorcom.com
  description:DAHDI OSLEC wrapper
  depends:dahdi
  vermagic:   2.6.26-1-486 mod_unload modversions 486
 
 
 
 
 
 
  2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com
 
  On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote:
  Hi,
  I've a problem: I can't configure DAHDI with ech canceller OSLEC.
  I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
  But when in /etc/dahdi/systems.conf I insert value
  echocanceller=oslec,1-4,
  command dahdi_cfg - give me an error about oslec.
  What is the output of:
 
   modinfo oslec
   modinfo dahdi_echocan_oslec
 
  --
Tzafrir Cohen
  icq#16849755  
  jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 jabber%3atzafrir.co...@xorcom.com jabber%253atzafrir.co...@xorcom.com
  +972-50-7952406   mailto:tzafrir.co...@xorcom.com
  http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
  ___

 I'm not sure that is strange. When I build DAHDI with OSLEC I don't get
 an oslec module, I get an echo module:

 r...@srvpbx:~# modinfo echo
 filename:   /lib/modules/2.6.27.19-smp/staging/echo/echo.ko
 version:0.3.0
 description:Open Source Line Echo Canceller
 author: David Rowe
 license:GPL
 srcversion: 285EC80D84DCE294A677160
 depends:
 vermagic:   2.6.27.19-smp SMP preempt mod_unload 686

 r...@srvpbx:~# modinfo dahdi_echocan_oslec
 filename:   /lib/modules/2.6.27.19-smp/dahdi/dahdi_echocan_oslec.ko
 license:GPL
 author: Tzafrir Cohen tzafrir.co...@xorcom.com
 description:DAHDI OSLEC wrapper
 depends:dahdi,echo
 vermagic:   2.6.27.19-smp SMP preempt mod_unload 686

 Try building DAHDI with the steps detailed here and see if you have
 better luck:

 http://lists.digium.com/pipermail/asterisk-users/2009-January/225299.html

 -Dave

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[asterisk-users] DAHDI with OSLEC

[Marco Sambo]

Hi,
I've a problem: I can't configure DAHDI with ech canceller OSLEC.
I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4,
command dahdi_cfg - give me an error about oslec.

Someone can help me?

Thanks

Marco
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Re: [asterisk-users] DAHDI with OSLEC

[Marco Sambo]

Mhmm. Thaht's strange!

modinfo oslec
--
modinfo: could not find module oslec

and

modinfo dahdi_echocan_oslec
--
filename:   /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
license:GPL
author: Tzafrir Cohen tzafrir.co...@xorcom.com
description:DAHDI OSLEC wrapper
depends:dahdi
vermagic:   2.6.26-1-486 mod_unload modversions 486






2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com

 On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote:
  Hi,
  I've a problem: I can't configure DAHDI with ech canceller OSLEC.
  I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
  But when in /etc/dahdi/systems.conf I insert value
 echocanceller=oslec,1-4,
  command dahdi_cfg - give me an error about oslec.

 What is the output of:

  modinfo oslec
  modinfo dahdi_echocan_oslec

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Ebay's SIP for Skype

[Marco Sambo]

I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?

Marco

2009/3/26 Grygoriy Dobrovolskyy megaho...@gmail.com

 skip2pbx is the best i tryed, but nasty price ;)

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Re: [asterisk-users] Ebay's SIP for Skype

[Marco Sambo]

Well,
anyone knows a good Skype vs SIP channel or program or something else to
integrate it into an Asterisk machine, to call normal skype users and not
and receive normal skype calls?
I red that Digium and Skype are working to integrate a chan_skype. Anyone
can tell me about?

Bye
Marco



2009/3/25 Administrator TOOTAI ad...@tootai.net

 Michael Robertson a écrit :
  Anyone connected up to it yet?
 
http://www.skypeforsip.com/
 
 
  This service is vaporware. It's just surveyware at this point with no
 actual
  service. An alternative is OpenSky which is a launched service which does
  SIP to Skype and Skype to SIP so you can answer and make all your Skype
  calls from any SIP aware device. There's a comparison chart at:
  http://sipforskype.com and you can learn more about the service at:
  http://gizmo5.com/opensky
 
 For us, opensky can be OK for individual users, not for allowing
 Asterisk users to call Skype users. Why? Simply that when you buy the 20
 USD connection to Skype and don't want your calls to be cutted after 5
 mn, you have to use the Gizmo Skype aliases system which is in your
 account. Not really helpful if you want to connect transparently your
 users to Skype! They better had to say Ok, this is your prefix (eg
 1333) to call Skype users through your account, this would allow
 us -as Asterisk admin- to format calls from *our* users in the right way.

 --
 Daniel

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Re: [asterisk-users] Busy on SIP

[Marco Sambo]

Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:

extensions.conf
===
exten = _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}1]?free:busy)
exten = _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten = _1X,n,Hangup()
exten = _1X,n(busy),Busy()
exten = _1X,n,Hangup()

sip.conf
===
[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=1
pickupgroup=1
subscribecontext=BLF_group
dtmfmode=info

[10](intphones)
context=IntPhones
username=10
secret=1234
amaflags=documentation
accountcode=sip10
callerid=sip10 10
call-limit=2
dial=SIP/10
canreinvite=no


And this resolve for me problems for busy and for xfer Aastra button.

Marco



2009/3/17 Ira i...@extrasensory.com

 At 01:29 AM 3/17/2009, you wrote:
 But there is another little problem. On Aastra phone (on other
 phones I don't try yet), the xfer button doesn't work, until I set
 call-limit=2, but making this I find the phone not busy.

 As far as I can tell on my Aastra phones it takes 2 links to complete
 a transfer. Pressing transfer puts the first call on hold and allows
 you to make a second call. Pressing transfer a second time then
 connects those to calls together and removes you from the call. If
 you only have 1 call allowed you'll need to implement that using
 features.conf or implement the busy stuff in the dial plan.

 Ira


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Re: [asterisk-users] 428 Loop Detected

[Marco Mouta]

It's so uncommon for me fxs and fxo cards and based on the reference
of sip.conf files and accounts i totally missed last paragraph where
it was mentioned only analogue lines and fxs (phone).

my appologies.

E1 and BRIs and sip trunks have been overloading my last month of work.

cheers,
--
Marco Mouta



2009/3/16 Steve Totaro stot...@totarotechnologies.com:
 Again, if I am interpreting this correctly, he is not using SIP.  A
 four port card 2fxo/2fxs means to me that he is not using SIP at all.

 If by card, you mean some kind of SIP gateway, then I misunderstood
 and the problem, but seeing DAHDI channels leads me to believe that
 SIP is not required and actually causing your problems.

 SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this
 case)...  If you had a SIP device, it would be connected to the data
 network, not a phone line.  Can you just plug your phone into a
 regular landline jack and get dialtone?  If so, forget SIP for now.

 Comment out or delete all your sip.conf peers since you are not using SIP.

 Change your dialplan to not (Dial/SIP but (Dial/DAHDI/1,10) and the
 correct channel to your FXS port that the phone is connected to.

 Thanks,
 Steve Totaro

 On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote:
 Hi,

 problem is that you are saying that phone in sip.conf is at the same
 ip address of your asterisk box so you are dialing into a loop to your
 self asterisk box

 [phone]
 type=friend
 context=phone1
 secret=g00dpazzwerd
 bindport=5060
 host=192.168.1.106
 dtmfmode=rfc2833

 what you need is:

 [phone]
 type=friend
 context=phone1
 secret=g00dpazzwerd
 dtmfmode=rfc2833
 host=dynamic
 ;configuring your codecs (i don't know what else you have configured,
 just preventing audio for you)
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm


 Dial sip/phone is enough too..

 [from-pstn]
 ;include = default
 exten = s,1,Dial(SIP/phone,10)
 exten = s,2,Voicemail(line)
 exten = s,3,Hangup


 hope it helps.

 don't forget to asterisk reload on cli.

 Looking forward to hearing from you.

 cheers

 --
 Marco Mouta



 On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote:
 Hi I looked at few emails related to this subject. And still not sure
 how to solve the loop detect problem for my case

 iqb...@improvise:/etc/asterisk$ cat sip.conf

 [general]
 context=line1

 [phone]
 type=friend
 context=phone1
 secret=g00dpazzwerd
 bindport=5060
 host=192.168.1.106
 dtmfmode=rfc2833

 [line]
 type=friend
 context=line1
 secret=anothers33cret
 bindport=5061
 host=192.168.1.106
 dtmfmode=rfc2833

 iqb...@improvise:/etc/asterisk$ cat extensions.conf
 [default]
 exten = s,1,Answer
 exten = s,2,Wait(2)
 exten = s,3,Playback(tt-monkeys)
 exten = s,4,Hangup

 [from-internal]
 include = default

 [phone1]

 [from-pstn]
 ;include = default
 exten = s,1,Dial(SIP/ph...@phone,10)
 exten = s,2,Voicemail(line)
 exten = s,3,Hangup

 [line1]


 So my home land line is going to the FXO port and my home phone is
 hanging off of FXS port.

 Here are the contexts for my fxo/fxs card


 improvise*CLI dahdi show channels
   Chan Extension  Context         Language   MOH Interpret
  pseudo            default                    default
      1            from-internal              default
      2            from-internal              default
      3            from-pstn                  default
      4            from-pstn                  default


 I want to call from my cell and make my home phone ring and if I dont
 pickup in 10 secs I want the call
 go to my voicemail. But I am getting a loop detect. The debug output
 is attached.

 What am I doing wrong?

 --
 Asif Iqbal
 PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu
 A: Because it messes up the order in which people normally read text.
 Q: Why is top-posting such a bad thing?



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 --
 Thanks,
 Steve Totaro
 +18887771888 (Toll Free)
 +12409381212 (Cell)
 +12024369784 (Skype)

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Re: [asterisk-users] Busy on SIP

[Marco Sambo]

Hi all,
maybe I find the problem and the solution.
I move the following parameters on section [general]:

[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
notifyringing=yes

and then on SIP account I put this:

[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=0
pickupgroup=0
dtmfmode=info

[10](intphones)
context=office
username=10
secret=1234
subscribecontext=BLF_group
call-limit=1


and this works!

When someone call SIP/10, and then I call again SIP/10, I find it busy.
On the other side, when SIP/10 make a call, and then I call again SIP/10, I
find it busy. And that's ok!

But there is another little problem. On Aastra phone (on other phones I
don't try yet), the xfer button doesn't work, until I set call-limit=2, but
making this I find the phone not busy.

Anyone know how to use busy-level parameter or some other useful parameters?


Thanks all

Marco


2009/3/16 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net


 On Mon, 16 Mar 2009, Olivier wrote:

  2009/3/16 Gordon Henderson
  gordon+aster...@drogon.net gordon%2baster...@drogon.net
 gordon%2baster...@drogon.net gordon%252baster...@drogon.net
 
 
  On Mon, 16 Mar 2009, Marco Sambo wrote:
 
  Hi,
  I have a question. How can I configure my sip.conf to make a SIP phone
  busy
  on incoming and outcoming calls? I explain my problem.
  When SIP phone receive a call and then I try to call that phone, I find
  it
  busy.
  When SIP phone make a call and I try to call that phone, I find it
  avaible
  and it rings but I want to find it busy.
 
  Disable call-waiting inside the phone.
 
  Doesn't call-limit=1 force the same behaviour ?

 It appears to limmit the number of outgoing calls from that phone and
 independantly the number of inoming calls.

 So a phone can make an outgoing call, and still take an incoming call, and
 vice-versa, with call-limit=1

 I also found early versions of this buggy in that it didn't seem to
 properly decrement the counter on hang-up, so is call-limit was set to 3,
 then that phone could only take 3 calls, one after the other, before it
 would be premenantly busyd, but this was a long time back, and it might
 have been something I was foing, but since then I always turned
 call-waiting off on the phones when users didn't want multiple call
 features.

 Gordon


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Re: [asterisk-users] Busy on SIP

[Marco Sambo]

Ok, I read it.

Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with
field CURCALLS.






2009/3/17 Philipp Kempgen philipp.kemp...@amooma.de

 Marco Sambo schrieb:
  Anyone know how to use busy-level parameter or some other useful
 parameters?

 call-limit=2
 busy-level=1
 ?

 busy-level is not in Asterisk 1.4 of course.


Philipp Kempgen
 --
 AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
 AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 --

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[asterisk-users] Busy on SIP

[Marco Sambo]

Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it rings but I want to find it busy.

I configure sip.conf like following:

[10]
type=friend
qualify=yes
host=dynamic
callgroup=0
pickupgroup=0
context=office
username=10
secret=1234
subscribecontext=BLF_group
limitonpeers=yes
call-limit=1
notifyringing=yes
dtmfmode=info


Someone can help me? I can't understand why I find it avaible when it makes
an outgoing call.

Thanks all

Marco
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Re: [asterisk-users] 428 Loop Detected

[Marco Mouta]

Hi,

problem is that you are saying that phone in sip.conf is at the same
ip address of your asterisk box so you are dialing into a loop to your
self asterisk box

[phone]
type=friend
context=phone1
secret=g00dpazzwerd
bindport=5060
host=192.168.1.106
dtmfmode=rfc2833

what you need is:

[phone]
type=friend
context=phone1
secret=g00dpazzwerd
dtmfmode=rfc2833
host=dynamic
;configuring your codecs (i don't know what else you have configured,
just preventing audio for you)
disallow=all
allow=ulaw
allow=alaw
allow=gsm


Dial sip/phone is enough too..

[from-pstn]
;include = default
exten = s,1,Dial(SIP/phone,10)
exten = s,2,Voicemail(line)
exten = s,3,Hangup


hope it helps.

don't forget to asterisk reload on cli.

Looking forward to hearing from you.

cheers

--
Marco Mouta



On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote:
 Hi I looked at few emails related to this subject. And still not sure
 how to solve the loop detect problem for my case

 iqb...@improvise:/etc/asterisk$ cat sip.conf

 [general]
 context=line1

 [phone]
 type=friend
 context=phone1
 secret=g00dpazzwerd
 bindport=5060
 host=192.168.1.106
 dtmfmode=rfc2833

 [line]
 type=friend
 context=line1
 secret=anothers33cret
 bindport=5061
 host=192.168.1.106
 dtmfmode=rfc2833

 iqb...@improvise:/etc/asterisk$ cat extensions.conf
 [default]
 exten = s,1,Answer
 exten = s,2,Wait(2)
 exten = s,3,Playback(tt-monkeys)
 exten = s,4,Hangup

 [from-internal]
 include = default

 [phone1]

 [from-pstn]
 ;include = default
 exten = s,1,Dial(SIP/ph...@phone,10)
 exten = s,2,Voicemail(line)
 exten = s,3,Hangup

 [line1]


 So my home land line is going to the FXO port and my home phone is
 hanging off of FXS port.

 Here are the contexts for my fxo/fxs card


 improvise*CLI dahdi show channels
   Chan Extension  Context         Language   MOH Interpret
  pseudo            default                    default
      1            from-internal              default
      2            from-internal              default
      3            from-pstn                  default
      4            from-pstn                  default


 I want to call from my cell and make my home phone ring and if I dont
 pickup in 10 secs I want the call
 go to my voicemail. But I am getting a loop detect. The debug output
 is attached.

 What am I doing wrong?

 --
 Asif Iqbal
 PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu
 A: Because it messes up the order in which people normally read text.
 Q: Why is top-posting such a bad thing?

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Re: [asterisk-users] Faxing success rate on PRI

[Marco Signorini]

Thank you, Doug, for precious information.

Best regards,
Marco Signorini.

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com


Doug Lytle wrote:

 Main fax server:


 Mandriva 2008.1
 Kernel 2.6.24.5 (Compiled for source)
 (1) Intel(R) Xeon(TM) CPU 2.80GHz
 Digium TE110P (23 b channels 1 data)

 Asterisk 1.4.20.1
 HylaFAX+ 5.2.7
 iaxmodem 1.2.0
 SpanDSP 0.0.4 (The one that came with iaxmodem)

 Doug

   


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Re: [asterisk-users] Faxing success rate on PRI

[Marco Signorini]

Hi Gordon, thank you for your answer.

It's not mandatory to use an external box to handle the PRI. I was
thinking to use a Patton device instead of a TE120P just because I would
like to be able to switch to T38 in the near future or if working with
inband faxes will reveal problems.
I'm open to any suggestion, being in the initial requirements analysis
stage.

External devices, also, let me able to select a server independently by
the PCI, PCIXpress or any other connectors will be ready in the future,
maximizing the customer investment... but this could be the less
interesting part.

Thank you to All People answered me on this subject.
Analyzing your answers, seems that fax handling is still today
problematic with IAXModem and Hylafax... or I'm wrong?
What about other solutions?

Thank you and best regards,
Marco Signorini

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com



Gordon Henderson wrote:
 On Sun, 8 Mar 2009, Marco wrote:

   
 Hi List,
 I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
 on my lab test setup and I appreciate it. Moreover the global quantity of
 fax handled by this setup is not very high.

 I'll be involved in a more complex system for a customer and I would like
 to ask to All of you if you have experiences and/or statistical results on
 faxing success and failure rate.

 The system I have to deploy will operate in the following context:

 - It will be interfaced to an E1 PRI
 - It will be able to send and receive faxes (by e-mail and/or virtual
 printers)
 - It will be able to send faxes from a normal fax machine.

 The system will be placed on the same building, i.e. only private ethernet
 trunks.

 I'm thinking to this type of solution:
 - Patton external unit for E1
 

 Out of curiosity, why an external box rather than something like a TE120P 
 PCI card?


   
 - Asterisk 1.4 + IAXModem + Hylafax
 - An external ATA for the fax machine
 but I'm open to any other possible solution (I'm thinking to have a
 demodulation on Patton and talk T38 with Asterisk 1.6).
 

 Personally, I think you're adding complexity and can't see why that would 
 be better than an on-board PRI card...

 Gordon
   


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Re: [asterisk-users] Faxing success rate on PRI

[Marco Signorini]

Thanks Doug and Lee,
your testimonials are changing my opinion :-)

Can you provide some details about your setup? What PRI solution are you
using? And what version of Asterisk, IAXModem, SpanDSP?

Thank you and best regards,
Marco Signorini

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com



Lee Howard wrote:
 Marco Signorini wrote:
   
 Analyzing your answers, seems that fax handling is still today
 problematic with IAXModem and Hylafax... or I'm wrong?
 

 A single server that I administer, receiving 12,000 pages and sending 
 1,000 pages daily would seem to contradict your conclusions.

 Thanks,

 Lee.
   


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Re: [asterisk-users] Faxing success rate on PRI

[Marco Signorini]

Hi Steve,
I was waiting for your answer :-P

I started to use your SpanDSP library since some years ago but,
unfortunately, my experience was only related to lab or personal use
and/or systems with PSTN or BRI cards and low fax volume where it's
impossible to have valid statistics.

I read the link you provided me and now I'm confident that IAXModem
running on the same asterisk box with a PRI board is something I can
propose to my customer.
There are some other variable I would like to evaluate, like, for
example, what type of PRI connection people of this list are using in
their fax servers.

Thank you for writing SpanDSP and best regards,
Marco Signorini

===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com


Steve Underwood wrote:
 Marco Signorini wrote:
   
 Thank you to All People answered me on this subject.
 Analyzing your answers, seems that fax handling is still today
 problematic with IAXModem and Hylafax... or I'm wrong?
 What about other solutions?
   
 
 I'm not sure where you got that idea. Most comments about iaxmodem + 
 HylaFAX are very positive. It does, of course, require a reliable 
 connection between iaxmodem and the PSTN, but most people set these 
 things up in a reasonably well controlled environment. There are some 
 notes at http://www.soft-switch.org/spandsp-soft-fax-performance.html 
 about the results of serious real world volume testing of iaxmodem.

 Regards,
 Steve

   


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[asterisk-users] Faxing success rate on PRI

[Marco]

Hi List,
I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
on my lab test setup and I appreciate it. Moreover the global quantity of
fax handled by this setup is not very high.

I'll be involved in a more complex system for a customer and I would like
to ask to All of you if you have experiences and/or statistical results on
faxing success and failure rate.

The system I have to deploy will operate in the following context:

- It will be interfaced to an E1 PRI
- It will be able to send and receive faxes (by e-mail and/or virtual
printers)
- It will be able to send faxes from a normal fax machine.

The system will be placed on the same building, i.e. only private ethernet
trunks.

I'm thinking to this type of solution:
- Patton external unit for E1
- Asterisk 1.4 + IAXModem + Hylafax
- An external ATA for the fax machine
but I'm open to any other possible solution (I'm thinking to have a
demodulation on Patton and talk T38 with Asterisk 1.6).

The fax volume will be high because actually the customer has a ZFax
software system with 12 fax-modem installed (that will be replaced by the
system).

I know that this was already asked in this list in the past, but I would
like to know if someone has experience on this and could share their
opinion, tricks and/or statistical results on failure/success rate when
faxing. I think that this could be useful to other people have to realize
a system like that one depicted.


Thank you in advance.
Marco Signorini

===
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http://www.ingegnitech.com



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Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

[Marco Signorini]

Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on SIP
channel (disabling the fax detection on SPA and sending fax inband) and
look at the recorded file with a wave editor (Audacity).
I had better results if the maximum level is near half to the full
dynamic. Then switch to T38, if you need it.


Hope this helps you.

Best regards,
Marco Signorini

===
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http://www.ingegnitech.com



Joseph wrote:
 On 03/04/09 19:31, Michael wrote:
   
 On Wed, 04 Mar 2009 19:25:38 Joseph wrote:
 
 I'm faxing from  stand alone fax machine via linksys SPA3102 but most of
 the time only half or quarter page goes through.

 Did anybody have any experience like this?
   
 Should be obvious but does your up line SIP provide support T.38?
 

 What do you mean it should be obvious?

 I think Linksys SPA3102 does support T.38 
 On Line 1 I have:
 FAX Enable T38: Yes
 FAX T38 Redundancy: 1
 FAX Passthru Codec: G711u
 FAX Process NSE: Yes
 FAX Passthru Method: NSE
 FAX CNG Detect Enable: Yes
 FAX CED Detect Enable: Yes

   


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Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

[Marco Signorini]

Joseph wrote:
 On 03/04/09 15:44, Marco Signorini wrote:
   
 Hi Joseph.
 I've spent some time tuning the SPA3102 FXS line input and output gain
 and I think that this is an important variable.
 Let's try to record incoming and outgoing fax tones with asterisk on SIP
 channel (disabling the fax detection on SPA and sending fax inband) and
 look at the recorded file with a wave editor (Audacity).
 I had better results if the maximum level is near half to the full
 dynamic. Then switch to T38, if you need it.
 

 As I remember I have experimented with gain on PSTN line as well but have 
 reset back to default.
 I have:
 SPA To PSTN Gain:0
 PSTN To SPA Gain:0

 I think 0  is the default.

   
Yes, 0 is the default.
Is the fax machine connected to the FXS port or do you use the SPA3102
only as a SIP 2 PSTN gateway?
If you use the FXS port, please take a look at the gain parameters you
can find in the Miscellaneous section in the Regional page (log in
as Administrator then switch to the advanced report).

Now I've -5 as input gain and -2 as output. I don't know if this could
helps you.


Best regards,
Marco Signorini


===
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http://www.ingegnitech.com


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[asterisk-users] patlooptest and TE121P

[Marco Signorini]

Hi List.
I'm running the patlooptest program I've found in dahdi_tools 2.1.0.2.
The target is a TE121P board with a loopback cable inserted on the
socket. I suppose that the loopback is working fine because I'm able to
see the green led on and dahdi_tool reports no errors.

When I run the patlooptest I've a lot of errors (the received values are
completely different than the transmitted) and I would like to know if
someone in the list had run this test with the TE121P board. This lets
me understand if the problem is on my board.

The setup I've is what I've found on http://kb.digium.com/entry/138/ for
the E1.

Thank you and best regards,

Marco Signorini

===
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http://www.ingegnitech.com

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Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Marco Signorini]

Hi Paulo,
It's right! I've changed the zend.zel_compatibility_mode to Off,
following your suggestion, and asterisk-stat is still working on PHP5.
Thank you!

Just for clarity: the default values for the two keys on OpenSuse 10.2
(updated to latest revision), and following, is Off.

Best regards,
Marco Signorini

===
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http://www.ingegnitech.com



Tiago Durante wrote:
 On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos paulo.r.san...@sapo.pt wrote:
   
 Marco Signorini wrote:
 
 Hi Tiago.

 I've it working on PHP 5.2.6 but only after having modified the php.ini
 default configuration keys:

 zend.ze1_compatibility_mode = Off
 short_open_tag = Off
   
 Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is
 set to On and it is working.

 Those are my defaults, at least I never changed them. Installed with
 apt-get on Debian 4.0, PHP version 5.2.0-8+etch13.
 

 Cool, I'm gonna test it and I let you guys know if worked or not.

 Thanks a lot!

   


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Re: [asterisk-users] building a phone

[Marco Signorini]

It's a dream!
It's since years that I'm thinking to have an open hardware project
targeted to a SIP application.

I'm thinking, for example, to have a modular system that can be targeted
to different custom appliances like, for example, (video) door bell
opener/intercom, or building/desktop music streamer, or SIP compliant
actuators.

I have a (very) little experience on electronic projects. Is there
something I can do to help starting a similar project?

Thank you and best regards.
Marco Signorini



Tzafrir Cohen wrote:
 Hi folks

 A common wisdom here is that one should use a proper hardware phone
 rather that an extra software on the user's PC. Why is that such a big
 issue?

 One thing that bothers me with the current crop of hardware SIP phones
 is that they are hopelessly properitary. 

 So what would it take to build a fully-adaptable phone?

 Here are some of my thoughts. This is not anything I plan to do soon (if
 at all), but I really find it strange that there aren't such phones
 already.


 == Small Quantities:
 When you look at such systems it becomes aparant that you can get much
 nicer prices if you buy large quanities. But this is something that will
 be a problem. Not only for prototying. The fact that you're limited to a
 strict hardware setting is very limiting. No mixing and matching like in
 a standard PC. I'm not exactly sure how to overcome that.

 == Platforms:
 There are many embedded platforms nowadays. I assume that the relevant
 application requires some non-trivial CPU power. I would exclude e.g. a
 486-based systems. My target phone should be able to handle at least two
 concurrent Speex calls. Preferrebly 6 speex calls and above.

 OTOH, I can't afford a monster CoreDuo. I need a quiet system with no
 fan. Thus the target CPU may be higher end VIA or Atom. Not sure about
 Geode. 

 There are also some interesting ARM-based boards around. I'm completely
 unfamiliar with them but I suspect that they may prove to be cheaper. 

 == SIP Software:
 Not really sure here. There must be something close to usable already, I
 guess.

 == Micro Browser:
 Hell no!

 The device should have an LCD display, and the content of that display
 should be programmable. Programming it using a HTML renderred is a bad
 design decision.

 The device should be a good phone. It should not attempt to be a web
 browser, as it will be a lousy one.

 == Handset:
 I suppose that an obvious starting point for a handset is skype phones
 such as USB handsets from yealink. Far from an optimal design, but a
 driver already exists.


 == Ease of Use:
 A phone must be usable. The target device must be something my mom can
 use. However that does not mean it must be easy to program. It must be
 programmable and hackable. But I can live with a complicated user
 interface for that. If such phones become successful and useful, better
 interfaces will eventually be written.


   


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Re: [asterisk-users] building a phone

[Marco Signorini]

Jon Pounder wrote:
 Marco Signorini wrote:
   
 It's a dream!
 It's since years that I'm thinking to have an open hardware project
 targeted to a SIP application.
   
 

 there is already a project called openmoko - join it and buy some hardware.

 The phone is large and clunky - the idea is good, but not something 
 you're ever going to carry in your pocket, and somewhat silly when there 
 is already smaller hardware out there that runs linux at less cost than 
 their device.

   

Thank you Jon,
Really interesting project! I'll follow it.

Best regards,
Marco Signorini



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Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

[Marco Signorini]

Hi Tiago.

I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:

zend.ze1_compatibility_mode = Off
short_open_tag = Off

setting together to On and restarting apache forces PHP5 to behave like
PHP 4.x version.

regards,
Marco Signorini

===
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http://www.ingegnitech.com



Paulo Santos wrote:
 Tiago Durante wrote:
   
 Hi all,

 I don't know if its the right place to ask, but... Does any one have
 the asterisk-stat-v2 running with PHP5?


 Tks!


 

 # php --version

 PHP 5.2.0-8+etch13 (cli) (built: Oct  2 2008 08:26:18)
 Copyright (c) 1997-2006 The PHP Group
 Zend Engine v2.2.0, Copyright (c) 1998-2006 Zend Technologies

 Working for me. Don't forget you need php5-gd for the graphics to show.

   


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Re: [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

[Marco Mouta]

try to set in your zapata.conf

overlapdial=yes

then in your asterisk cli

reload chan_zap.so


--
Marco Mouta



On Fri, Feb 13, 2009 at 9:21 AM,  joek...@gmail.com wrote:
 Default FreePBX context,

 [from-pstn]
 include = from-pstn-custom ; create this context in
 extensions_custom.conf to include customizations
 include = ext-did
 include = ext-did-post-custom
 include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
 enabled, should be bofore ext-local
 include = ext-did-catchall; THIS MUST COME AFTER ext-did
 exten = fax,1,Goto(ext-fax,in_fax,1)

 The call seems to be going here

 [ext-did-catchall]
 include = ext-did-catchall-custom
 exten = s,1,Noop(No DID or CID Match)
 exten = s,n(a2),Answer
 exten = s,n,Wait(2)
 exten = s,n,Playback(ss-noservice)
 exten = s,n,SayAlpha(${FROM_DID})
 exten = s,n,Hangup
 exten = _.,1,Set(__FROM_DID=${EXTEN})
 exten = _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
 exten = _.,n,Goto(s,a2)
 exten = h,1,Hangup

 ; end of [ext-did-catchall]

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[asterisk-users] Suggestion for a new server for E1 line

[Marco Signorini]

Hi All,

I'm trying to identify a new server as a replacement for what our
customer actually has (DELL PowerEdge 860).
The server will mount the Digium board TE121, we already have, with echo
cancel onboard. I need to know if someone could suggest a new server
that's compatible with this board. With compatible I mean that's not
having any problem like IRQ sharing, IRQ miss or kernel panic with
DAHDI/Zaptel drivers or unwanted hangup or noise during the conversation.

My other needs are:

1. At least two RAID disk (preferably hot-swappable but I'm looking also
for solution without this feature)
2. Possibly with redundant power supply
3. 1U or 2U size
4. As cheap as possible.

Our customer is pushing to have the HP Proliant DL120 but I think it's
not fitting the 24/7 needs it has. The server will be used to dispatch
calls coming from an 800 free number for one humanitarian organization
in Italy.

Any suggestion is really welcomed.
Thank you very much.

Best regards,
Marco Signorini


http://www.ingegnitech.com



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Re: [asterisk-users] Dahdi Init script for Suse?

[Marco]

Hi,
I've it up and running on OpenSuse 11. I used the scripts provided by the
sources and commented out one line:

#
# Determine which kind of configuration we're using
#
#system=redhat  # assume redhat
system=debian # assume debian

This forces the script to use debian style. It works for me, except, if I
remember well, some little problem on reload (but stopping and starting
again works fine).

Best regards,
Marco Signorini.

==
INGEGNI Tech S.r.l.
http://www.ingegnitech.com


 Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box
 that'll work right?  The one included by default only deals with debian
 and redhat, and the changes between the old zaptel script I have that
 works are far too invasive.  Notably in the use of this action command
 that's probably redhat specific.

 There's practically zilch on google on the matter.  I think suse support
 should be included by default, though.

 Thanks!,

 Joshua Kinard
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Re: [asterisk-users] Portech MV-378 with Asterisk

[Marco Signorini]

Yes.
That's the correct way to do it. Placing # as a rule in callnum forces
the Portech to use the number defined in the SIP INVITE packet.

Bye.
Marco.


Marco Signorini
INGEGNI Tech S.r.l.
http://www.ingegnitech.com http://www.ingegnitechcom/


Pascal Bruno wrote:
 Sorry for bothering you, but I got it, I just had to put # in callnum!



 On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno tipas...@gmail.com
 mailto:tipas...@gmail.com wrote:

 I want to dial out using the sim card.  What I did, I have used
 the SIP channel ex:

 Channel: SIP/thenum...@mv378

 It shows the called is being made in the dialplan, but the number
 I have entered does not dial, it just goes straight to the
 specified dialplan extensions.

 Then what I did, in the Lan to Mobile Table, I put * in url and
 the number I wanted to dial in call num, then the call was made to
 that number using the sim card properly.

 I was wondering if I cannot supply the number to be dialed using
 an asterisk call file, or do I have to put that number in the Lan
 to Mobile table.

 Any help would be appreciated.

 Thanks





 On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno tipas...@gmail.com
 mailto:tipas...@gmail.com wrote:

 Marco,

 The configs work fine for me.  I can receive calls with no
 problem.  Now, were you able to dial using the sim card?  I
 cant figure out how I can do it since asterisk doesnt have a
 channel to place call through the portech gateway.




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Re: [asterisk-users] Portech MV-378 with Asterisk

[Marco Signorini]

Emmanuel Pascal Bruno wrote:
 Has anyone been able to configure portech's mv-378 gateway with asterisk?
  
 I did the configuration as per the manual but it does not work.
  
 My server sees the portech gateway, but when the gateway is trying to
 register to my server it fails.  It says peer is not suppose to register.
  
 The gateway and the asterisk box are on two different location (two
 network, 2 differrent IP address).
  
 I would appreciate any kind of tutorial or advice on how to make it work.
  
 Thanks
 

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Hi,
I've an installation working with Portech MV-370. I'm supposing it's
quite similar to what you have. If it could be useful to you, this is my
sip.conf configuration file.

[GSMGtw1]
type=friend
context=from-gsm
host=dynamic; we have a DHCP assigned address
secret=reallyverysecret
nat=no  ; there is not NAT between phone and
Asterisk
canreinvite=no
dtmfmode=INFO
insecure=invite ; required to overcome authentication
problems in incoming calls
call-limit=1   ; permit only 1 outgoing call at a time
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=500

I remember that I've found a bug on the firmware that prevents to the
unit to register correctly on my asterisk box unless I'm using the raw
IP address instead of the name of the asterisk box. I remember something
wrong in cryptography chiper/dechiper based on realm... So, if you have
problems, let's try to specify the asterisk raw IP address in the Portech.

Best regards,
Marco Signorini.

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Re: [asterisk-users] Portech MV-378 with Asterisk

[Marco Signorini]

Pascal Bruno wrote:
 Thanks for your reply!
  
 Can you tell me what you have in your Portech configuration settings
 (Mobile to Lan Settings; Sip Proxy settings etc...)  My sip.conf file
 is pretty similar to yours but still cant register.


  
 On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini marcota...@libero.it
 mailto:marcota...@libero.it wrote:

 Emmanuel Pascal Bruno wrote:
 Has anyone been able to configure portech's mv-378 gateway with
 asterisk?
  
 I did the configuration as per the manual but it does not work.
  
 My server sees the portech gateway, but when the gateway is
 trying to register to my server it fails.  It says peer is not
 suppose to register.
  
 The gateway and the asterisk box are on two different location
 (two network, 2 differrent IP address).
  
 I would appreciate any kind of tutorial or advice on how to make
 it work.
  
 Thanks
 

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 Hi,
 I've an installation working with Portech MV-370. I'm supposing
 it's quite similar to what you have. If it could be useful to you,
 this is my sip.conf configuration file.

 [GSMGtw1]
 type=friend
 context=from-gsm
 host=dynamic; we have a DHCP assigned address
 secret=reallyverysecret
 nat=no  ; there is not NAT between phone
 and Asterisk
 canreinvite=no
 dtmfmode=INFO
 insecure=invite ; required to overcome
 authentication problems in incoming calls
 call-limit=1   ; permit only 1 outgoing call
 at a time
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 qualify=500

 I remember that I've found a bug on the firmware that prevents to
 the unit to register correctly on my asterisk box unless I'm using
 the raw IP address instead of the name of the asterisk box. I
 remember something wrong in cryptography chiper/dechiper based on
 realm... So, if you have problems, let's try to specify the
 asterisk raw IP address in the Portech.

 Best regards,
 Marco Signorini.



Hi,

I don't know if the problem could be in the Mobile to Lan or Lan to
Mobile settings because these  settings are related on how calls coming
from/to mobile are routed.  I didn't use the Portech routing features at
all because I need a simple GSM gateway to/from the asterisk box.
For this reason:
1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5
where mob is the extension I've generated in the asterisk box under
the context where the Portech operates;
2. The only rule I've on Lan to Mobile is URL=*; Call Num=#

I think the most relevant parameters for your problem are under the
Service Domain menu option (assuming that the firmware you have is
similar to what I've). On this menu I've compiled the 1st Realm (as I've
only one account) like that:

UserName: GSMGtw1
RegisterName: GSMGtw1
RegisterPassword: reallyverysecret
Domain Server: 192.168.0.5
Proxy Server: 192.168.0.5

Pay attention that, having specified the Domain Server with the raw IP
address, asterisk needs to be able to authenticate peers associated to
that. For this reason I've set:

domain=192.168.0.5

on sip.conf [general] section (remember to issue a sip reload from
asterisk cli).

Hope this helps!


Best regards.
Marco Signorini




Marco Signorini
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
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Re: [asterisk-users] Oslec issue

[Marco Signorini]

Ok Joseph. Don't worry, take your time :-)

For what's concerning the quality: I can assume my phone line is an
exception because it has a lot of echo. I've spent a LOT of time trying
to have an SPA3102 and an HT488 working without any reasonable result.
I'm playing for fun with zaptel/dahdi ec's since years and I was never
able to have a satisfying result with any ec provided with it. Neither
the fxotune process, neither any Tx/Rx gain or echo training parameter
tuning, neither Digium people that connected to my server 3 or 4 years
ago were able to completely solve any echo issue.

Some years ago I had the opportunity to test on my line HPEC with a
customer's box equipped with a TDM400P and I was impressed by the quality.

I think OSLEC is a very good piece of code. It's working fine with my
line and my Grandstream phones and, the must important thing, it's open
and free to use. Sometimes I can ear some echo or strange effects at the
very beginning of a call, but this is something that I can accept. In
the past I tried to modify the zaptel sources in order to prevent them
to free the oslec instance at each call. I think that my mods were not
working on systems where more than one zap channel was present and I was
not able to test it on these type of situations.

Thank you and bye
Marco Signorini





Joseph L. Casale wrote:
 I spent some time to understand what's missing in the OSLEC patch for
 dahdi... I can confirm the same problem you reported some days ago and I
 need OSLEC for home personal use.
 

 Wow,
 Appreciate the info! I will need a few days to get this done. Out of 
 curiosity,
 how do you find this ec's quality compared to the shipped modules and hpec?

 Thanks!
 jlc
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Re: [asterisk-users] Oslec issue

[Marco Signorini]

Hi Joseph.
I spent some time to understand what's missing in the OSLEC patch for
dahdi... I can confirm the same problem you reported some days ago and I
need OSLEC for home personal use.

For what I've understood looking at the code, there is some missing in
the dahdi_echocan_oslec.c file you can find in the
dahdi-linux/drivers/dahdi. I can list below what I did to have it
working. Actually I'm using the trunk revision 5443.

1. In the function echo_can_create i've modified the line

*ec = (struct echo_can_state *)oslec_create(ecp-tap_length,0);

with

*ec = (struct echo_can_state *)oslec_create(ecp-tap_length,
ECHO_CAN_USE_ADAPTION | ECHO_CAN_USE_NLP  | ECHO_CAN_USE_CLIP |
ECHO_CAN_USE_TX_HPF | ECHO_CAN_USE_RX_HPF);

This instructs OSLEC to have a working modality properly set.


2. I've replaced the function echo_can_update with the code below:

static void echo_can_update(struct echo_can_state *ec, short *iref,
short *isig)
{
unsigned int SampleNum;

for (SampleNum = 0; SampleNum  DAHDI_CHUNKSIZE; SampleNum++,
iref++)
{
short iCleanSample;
iCleanSample = (short) oslec_update((struct oslec_state
*)ec, *iref, *isig);
*isig++ = iCleanSample;
}
}

This lets the OSLEC to work on complete DAHDI_CHUNKSIZE buffer.


Please, if you have time, let me know if this solves your problem and,
if yes, I'll appreciate to have it public on trunk. I never did a commit
on asterisk svn so I need some hints on how to do it.


Thank you and best regards.
Marco Signorini.



Joseph L. Casale wrote:
 Yesterday I pulled in the latest svn of Dahdi and added the files
 from a recent kernel in the drivers/staging/echo structure and modified
 the Kbuild file so it would compile without error. I insmod'ed the module
 in, and modified my system.conf has echocanceller=oslec.

 cat /proc/dahdi/1 shows:
 Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER)
 IRQ misses: 1

1 WCTDM/0/0 FXSKS (In use)  (EC: OSLEC)
2 WCTDM/0/1
3 WCTDM/0/2
4 WCTDM/0/3

 With the reco's from http://www.rowetel.com/ucasterisk/oslec.html#install on
 configuring the chan_dahdi.conf file, the system behaves exactly as if there
 is no ec enabled at all?

 Are there any additional steps needed to enable oslec under dahdi, I am 
 guessing
 I have missed something?

 Thanks,
 jlc

   


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[asterisk-users] Persistentmembers (Not working with restart)

[Cordeiro, Marco]

Hello All,

 

I currently have an Asterisk Box, running a callcenter with 04 queues. I set
queues.conf with persistentmembers=yes in the general section as follows:

 

[general]

monitor-type = MixMonitor

persistentmembers = yes

 

However when I perform any kind of restart in the Asterisk application, all
agents are considered unavailable after that. 

 

Though when performing reload, agents keep their status as it was before the
reload. 

 

Is there any where else that I should set dynamic agents as persistent
members to keep their status after a asterisk restart??

 

Thanks,

 

Marco

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Re: [asterisk-users] Problem with DAHDI and OSLEC integration.

[Marco Signorini]

Hi Joseph and Tzafrir.
Thank you for your suggestions, feedbacks.

For Joseph: yes. I had the same warning messages and I solved with the
trick I suggested. Now oslec seems working (or at least and I can set it
through the dahdi_cfg command ;-) ).

For Tzafrir: here are the steps I did:
1. Taken the svn revision 5366 into my temporary folder
/home/marco/Install/dahdi-linux
2. Taken the linux-2.6.27 kernel sources baseline and placed in my
temporary folder /home/marco/install/linux-2.6.27
3. Taken the Linux kernel patch-2.6.28-rc6.gz, unzipped and applied to
the baseline kernel 2.6.27. This generates the folder
...linux-2.6.27/drivers/staging/echo
4. Copied the folder /staging/echo into
/home/marco/Install/dahdi-linux/drivers
5. Uncommented the oslec related two lines in the file Kbuild
6. From the folder /home/marco/install/dahdi-linx I've issued the
command make

The compiler starts and seems not able to compile what's present in the
folder /home/marco/Install/dahdi-linux/drivers/staging/echo. This
produces the warning already reported by Joseph and the inability to run
the oslec module.
I've had better results modifying the line:

obj-m += ../staging/echo/

with

obj-m += ../staging/echo/echo.o


in the Kbuild file.
I don't know if could be helpful, but I'm running these stuffs on
OpenSuse 11.

Thank you and best regards,
Marco Signorini.



Joseph L. Casale wrote:
 Have you copied there the files from the directory drivers/staging/echo
 in a recent (that is: = 2.6.28-rc1) kernel tree?
 

 Tzafrir,
 Thank you for following up on this. I don't have a quick command for only
 the three files, I just grabbed the tar ball. But like the OP, the only
 difference was that he used 2.6.28-rc6 and I used 2.6.28-rc5. I am pretty
 sure we had the same errors which I posted:
 http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html

 Thanks for any pointers!
 jlc

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