[asterisk-users] strange things on call transfer

[Christoph Fürstaller]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

I'm setting up an Asterisk system which is connected to an Alcatel 4400
PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
call by hitting the # key, I get this messages and nothing happens on
the phone:

WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame
that isn't a multiple of 50 bytes long from RTP (4)?

But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I
tet this error and the call is hung up, when transfering:

Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
find a codec translation path from ilbc to slin
Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
find a codec translation path from ilbc to slin
Mar 14 09:04:58 WARNING[30436]: channel.c:2752
ast_channel_make_compatible: No path to translate from Zap/31-1(72) to
SIP/374-08199e50(1024)
Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't
make Zap/31-1 and SIP/374-08199e50 compatible
Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call:
Bridge failed on channels Zap/31-1 and SIP/374-08199e50

Attached there is a SIP debug from such a call (with ilbc loaded)

I hope someone can help me.

chris...
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<-- SIP read from 172.28.20.4:2051:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" ;tag=48rd3g03e1
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.2.3
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 1775117380 1775117380 IN IP4 172.28.20.4
s=call
c=IN IP4 172.28.20.4
t=0 0
m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 
inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

--- (18 headers 19 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 172.28.20.4 : 2051 (NAT)
Reliably Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4
From: "Test User3" ;tag=48rd3g03e1
To: ;tag=as49d79f99
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user '104'
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" ;tag=48rd3g03e1
To: ;tag=as49d79f99
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Max-Forwards: 70
Contact: ;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport
From: "Test User3" ;tag=48rd3g03e1
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.2.3
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest 
username="104",realm="xxx.xx",nonce="0986769d",uri="sip:[EMAIL 
PROTECTED];user=phone",response="0e5cb3b8871eff4c6670a6f277ef3594",algorithm=md5
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 1775117380 1775117380 IN IP4 172.28.20.4
s=call
c=IN IP4 172.28.20.4
t=0 0
m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 
inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

--- (19 headers 19 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 172.28.20.4 : 2051 (NAT)
Found user '104'
Found RTP audio

RE: [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking

[Biju]

Have you added this line in your sip.onf ?

videosupport=yes 

Biju

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benedikt Franz
Sent: Wednesday, March 14, 2007 10:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisknow with video and X-Lite not
quiteworking

Hi Dave,

yes, Audio is fine, but no video. And as far as I can see, X-Lite (running
3.0 build 34025 here, all Clients have exactly the same version) supports
only the h.263 and h.263p video codecs. But I am not quite sure if I enabled
these codecs properly. For *now, I have put the allow-lines into the
users.conf, for instance, heres my setup (I cencored out email and secret):

[6510]
fullname = Benedikt Franz
secret = ...
email = ...
cid_number = 6510
zapchan =
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = yes
hasmanager = yes
callwaiting = yes
threewaycalling = yes
mailbox = 6510
hasagent = no
group =
host = dynamic
registersip = yes
registeriax = yes
allow = h263
allow = h263p
canreinvite = yes


I am not sure if that is a NAT problem, since all users are either on the
local area network, or connected through VPN (I have not tested video with
those yet, though), however, I will try that.
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Re: [asterisk-users] voicemail scenario

[richard Coco]

Hi,

some additional informations what i am trying to do.
In the voicemail.conf you have several setting for the
[general] section. One is the emailsubject. I have
something like emailsubject=New voicemail for
${VM_NAME}. In my [contexts] i have.

[context_section] 
extension_number =>
voicemail_password,user_name,user_email_address,user_pager_email_address,user_option(s)

  
The user_option(s) field can be used to override
default settings defined in the general section. There
are nine settings which may be used... unfortunately
not emailsubject. My question is, is there an
alternative way to override the default setting for
emailsubject defined in the [general]

thx.

--- Dovid B <[EMAIL PROTECTED]> wrote:

> I dont think you can but you can use a variable.
> Have a look at 
> voicemail.conf. You can edit the message the
> asterisk sends out. If you want 
> the CID to be in the subject you can use the
> variable ${CALLERID(number)} .
> 
> - Original Message - 
> From: "richard Coco" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion" 
> 
> Sent: Tuesday, March 13, 2007 10:53 PM
> Subject: Re: [asterisk-users] voicemail scenario
> 
> 
> 
> Hi,
> 
> i finally managed to get it work using GlobalVar.
> I still have a question. I have several context in
> my
> voicemail.conf like
> [default]
> [customer_1]
> [customer_2]
> [customer_3]
> 
> How can i set a different "emailsubject" for each
> context?
> 
> thx
> 
> 
> --- richard Coco <[EMAIL PROTECTED]> wrote:
> 
> > Hi all,
> >
> > i need help to implement a voicemail scenario.
> What
> > i
> > am trying to do is the following.
> >
> > user X dials a direct access for user Y voicemail
> > and
> > is asked to enter a number (e.g 12345678) and then
> > leaves a message. Then asterisk sends a
> notification
> > with attachement. The problem is that i need the
> > number entered (e.g 12345678) in the subject. Is
> > that
> > possible.
> >
> > thx in advance.
> >
> >
> >
> >
>

> > Don't pick lemons.
> > See all the new 2007 cars at Yahoo! Autos.
> > http://autos.yahoo.com/new_cars.html
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> >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> 
>

> The fish are biting.
> Get more visitors on your site using Yahoo! Search
> Marketing.
>
http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php
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> 



 

Don't get soaked.  Take a quick peek at the forecast
with the Yahoo! Search weather shortcut.
http://tools.search.yahoo.com/shortcuts/#loc_weather
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[asterisk-users] TDM-400, Polycom SIP phones, and echo problems

[Peter Doyle]

Hi Stephen,

I'm sure someone here can help with the echo, but I just wanted to say I
can highly recommend the Sangoma a200d boards with Octasic hardware echo
cans.  I spent a couple months trying to deal with echo using a TDM400p
and even a Vegastream SIP gateway with hardware echo can.  The quality
of the Octasic echo can on the Sangoma kit is amazing- literally 5
minutes after installing my a200d my echo was gone.  Once it has
converged (which happens very quickly), its just like a "normal" phone.
With the Vega most of the echo was gone due to the echo can, but we saw
echo in "double-talk" situations.  

 

It was more expensive, but definitely worth it.  

 

I would definitely also look into the new HPEC or Sangoma Softecho,
which I just saw on the wiki (voip-info.org).  Might be a good option,
though I haven't tried either (there was a discussion on this list about
the HPEC canceller just a few days ago)

Pete

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Re: [asterisk-users] strange things on call transfer

[MIVOC Systemes SAS]

Hello,

Can you  tell about  your configuration ( connection between Alacaltel 4400
and Asterisk ?)  : what  hardware ? what config files ?

Thank you

Minh VO
MIVOC Systemes SAS
FRANCE
- Original Message -
From: "Christoph Fürstaller" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, March 14, 2007 9:16 AM
Subject: [asterisk-users] strange things on call transfer


> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> I'm setting up an Asterisk system which is connected to an Alcatel 4400
> PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
> call by hitting the # key, I get this messages and nothing happens on
> the phone:
>
> WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame
> that isn't a multiple of 50 bytes long from RTP (4)?
>
> But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I
> tet this error and the call is hung up, when transfering:
>
> Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> find a codec translation path from ilbc to slin
> Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> find a codec translation path from ilbc to slin
> Mar 14 09:04:58 WARNING[30436]: channel.c:2752
> ast_channel_make_compatible: No path to translate from Zap/31-1(72) to
> SIP/374-08199e50(1024)
> Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't
> make Zap/31-1 and SIP/374-08199e50 compatible
> Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call:
> Bridge failed on channels Zap/31-1 and SIP/374-08199e50
>
> Attached there is a SIP debug from such a call (with ilbc loaded)
>
> I hope someone can help me.
>
> chris...
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v2.0.3 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF969jR0exH8dhr/YRAgP0AJ94ygGEPYHtGvLS7McUTrRAP1IkCgCgozv6
> rfuVGufsb8wQT3Iwl0ipXNg=
> =hrcE
> -END PGP SIGNATURE-
>
> --
-
> Orange vous informe que cet  e-mail a ete controle par l'anti-virus mail.
> Aucun virus connu a ce jour par nos services n'a ete detecte.
>
>






> <-- SIP read from 172.28.20.4:2051:
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
> From: "Test User3" ;tag=48rd3g03e1
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: ;flow-id=1
> P-Key-Flags: keys="3"
> User-Agent: snom300/6.2.3
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Supported: timer, 100rel, replaces, callerid
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 473
>
> v=0
> o=root 1775117380 1775117380 IN IP4 172.28.20.4
> s=call
> c=IN IP4 172.28.20.4
> t=0 0
> m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:2 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=encryption:optional
> a=sendrecv
>
> --- (18 headers 19 lines) ---
> Using INVITE request as basis request -
[EMAIL PROTECTED]
> Sending to 172.28.20.4 : 2051 (NAT)
> Reliably Transmitting (no NAT) to 172.28.20.4:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4
> From: "Test User3" ;tag=48rd3g03e1
> To: ;tag=as49d79f99
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
> Found user '104'
> salxpbx1*CLI>
> <-- SIP read from 172.28.20.4:2051:
> ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
> From: "Test User3" ;tag=48rd3g03e1
> To: ;tag=as49d79f99
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 ACK
> Max-Forwards: 70
> Contact: ;flow-id=1
> Content-Length: 0
>
>
> --- (9 headers 0 lines) ---
> salxpbx1*CLI>
> <-- SIP read from 172.28.20.4:2051:
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport
> From: "Test User3" ;tag=48rd3g03e1
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> Max-Forwards: 70
> Contact: ;flow-id=1
> P-Key-Flags: keys="3"
> User-Agent: snom300/6.2.3
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
> Allow-Events: talk, hold, refe

[asterisk-users] beronet BN4S0

[Thomas Stein]

Hello.

Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line.
misdnportinfo gives (what does ":Layer 4 protocol 0x0401 is detected, but 
not allowed for TE lib" mean?):

best regards and thanks
t.

asterix asterisk # misdnportinfo

Port  1: TE-mode BRI S/T interface line (for phone lines)
 -> Protocol: DSS1 (Euro ISDN)
 -> Layer 4 protocol 0x0401 is detected, but not allowed for TE lib.
 -> childcnt: 2
 * Port NOT useable for PBX (maybe there is already a PBX running?)

Port  2: TE-mode BRI S/T interface line (for phone lines)
 -> Protocol: DSS1 (Euro ISDN)
 -> Layer 4 protocol 0x0401 is detected, but not allowed for TE lib.
 -> childcnt: 2
 * Port NOT useable for PBX (maybe there is already a PBX running?)

Port  3: TE-mode BRI S/T interface line (for phone lines)
 -> Protocol: DSS1 (Euro ISDN)
 -> Layer 4 protocol 0x0401 is detected, but not allowed for TE lib.
 -> childcnt: 2
 * Port NOT useable for PBX (maybe there is already a PBX running?)

Port  4: TE-mode BRI S/T interface line (for phone lines)
 -> Protocol: DSS1 (Euro ISDN)
 -> Layer 4 protocol 0x0401 is detected, but not allowed for TE lib.
 -> childcnt: 2
 * Port NOT useable for PBX (maybe there is already a PBX running?)


mISDN_close: fid(3) isize(131072) inbuf(0x804c060) irp(0x804c060) 
iend(0x804c060)
-- 
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Re: [asterisk-users] beronet BN4S0

[Massimo Nuvoli]

Thomas Stein ha scritto:
> Hello.
> 
> Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line.
> misdnportinfo gives (what does ":Layer 4 protocol 0x0401 is detected, but 
> not allowed for TE lib" mean?):
> 

This is normal, the "channel" is used by asterisk and so is not
available to che misdnportinfo.

-> * Port NOT useable for PBX (maybe there is already a PBX running?)

Stop asterisk and do the same command!

Bye.
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org:Progetto Archivio SRL
adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia
email;internet:[EMAIL PROTECTED]
title:Amministratore Delegato
tel;work:0121303544
tel;fax:0121040601
x-mozilla-html:FALSE
url:www.progettoarchivio.com
version:2.1
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[asterisk-users] ChanSpy with Record() : Doesnt seem to work for an unbridged SIP call

[Onglipo Chengoli]

Hello,

I have an incoming Voip call, where an intro message is played (How can I 
help you today?) and the answer is recorded - with Record(tst:gsm,3,30). The 
call is not bridged. I also have an internal SIP phone which dials into an 
extension as soon as the Voip call lands, and spies on the unbridged Voip 
call, using ChanSpy.


All is well before start of Record () - ChanSpy can hear the incoming Voip 
line. But as soon as Record starts, ChanSpy audio gets cutout. The debug log 
has :
channel.c: Spy 'ChanSpy' on channel 'SIP/1-08cea3e0' read queue too 
long, dropping frames


Things are back to normal when Record finishes.

Any suggestions on a solution? Any insights into why the spy queues are not 
being emptied during Record?


Thanks for your help...
-Ong

_
Palate teasers: Straight from Master Chef Sanjeev Kapoor 
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Re: [asterisk-users] strange things on call transfer

[Christoph Fürstaller]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

I think I solved the 'problem'. The problem is my phone, Thomson
ST2030S. I set the DTMF Mode to RFC2833, as I usually do. On the Thomson
phone, I get this error when I try to hit/send a number or #. If I set
DTMF Mode to SIP INFO, it works. the interesting thing is, when I call
voicemailbox (with DTMF Mode RFC2833) I can login to the
voicemailsystem, but if I hit # during a call, to transfer it, I get
warnings and the transfer doesn't work. Seemes like this is a bug or bad
implementation in the phone?

My setup:
Asterisk (NT) is connected via 1xE1 (Sangoma A104) to Alcatel 4400 PBX
(TE). Asterisk gets clock from Alcatel. Do you need that to provide help
or are you interested how I did it?

chris...

MIVOC Systemes SAS schrieb:
> Hello,
> 
> Can you  tell about  your configuration ( connection between Alacaltel 4400
> and Asterisk ?)  : what  hardware ? what config files ?
> 
> Thank you
> 
> Minh VO
> MIVOC Systemes SAS
> FRANCE
> - Original Message -
> From: "Christoph Fürstaller" <[EMAIL PROTECTED]>
> To: 
> Sent: Wednesday, March 14, 2007 9:16 AM
> Subject: [asterisk-users] strange things on call transfer
> 
> 
> Hi,
> 
> I'm setting up an Asterisk system which is connected to an Alcatel 4400
> PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
> call by hitting the # key, I get this messages and nothing happens on
> the phone:
> 
> WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame
> that isn't a multiple of 50 bytes long from RTP (4)?
> 
> But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I
> tet this error and the call is hung up, when transfering:
> 
> Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> find a codec translation path from ilbc to slin
> Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> find a codec translation path from ilbc to slin
> Mar 14 09:04:58 WARNING[30436]: channel.c:2752
> ast_channel_make_compatible: No path to translate from Zap/31-1(72) to
> SIP/374-08199e50(1024)
> Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't
> make Zap/31-1 and SIP/374-08199e50 compatible
> Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call:
> Bridge failed on channels Zap/31-1 and SIP/374-08199e50
> 
> Attached there is a SIP debug from such a call (with ilbc loaded)
> 
> I hope someone can help me.
> 
> chris...
>>
- --
> -
Orange vous informe que cet  e-mail a ete controle par l'anti-virus mail.
Aucun virus connu a ce jour par nos services n'a ete detecte.
>>
>>

> 
> 


<-- SIP read from 172.28.20.4:2051:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" ;tag=48rd3g03e1
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.2.3
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473
>>
v=0
o=root 1775117380 1775117380 IN IP4 172.28.20.4
s=call
c=IN IP4 172.28.20.4
t=0 0
m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
>>
- --- (18 headers 19 lines) ---
Using INVITE request as basis request -
> [EMAIL PROTECTED]
Sending to 172.28.20.4 : 2051 (NAT)
Reliably Transmitting (no NAT) to 172.28.20.4:2051:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
> 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4
From: "Test User3" ;tag=48rd3g03e1
To: ;tag=as49d79f99
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d"
Content-Length: 0
>>
>>
- ---
Scheduling destruction of call
> '[EMAIL PROTECTED]' in 15000 ms
Found user '104'
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
From: "Test User3" ;tag=48rd3g03e1
To: ;tag=as49d79f99
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Max-Forwards: 70
Contact: ;flow-id=1
Content-Length: 0
>>
>>
- --- (9 headers 0 lines) ---
salxpbx1*CLI>
<-- SIP read from 172.28.20.4:2051:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via:

Re: [asterisk-users] strange things on call transfer

[MIVOC Systemes SAS]

Hi Chris,

I have the same problem because the pound(#) transfer with a Matra PABX.
I search a solution to replace the # touch that cause many problems
In France, we use the R touch to call transfer ...
In any way , I'm interest in your setup, your problems and your solutions
because we have a same project
( Different PABX Alacatel and Matracom  !! )
( I have a ASUS-ISDN Card in the Asterisk Box  Centos 4.4 )


Minh VO
MIVOC Systemes



- Original Message -
From: "Christoph Fürstaller" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, March 14, 2007 10:44 AM
Subject: Re: [asterisk-users] strange things on call transfer


> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> I think I solved the 'problem'. The problem is my phone, Thomson
> ST2030S. I set the DTMF Mode to RFC2833, as I usually do. On the Thomson
> phone, I get this error when I try to hit/send a number or #. If I set
> DTMF Mode to SIP INFO, it works. the interesting thing is, when I call
> voicemailbox (with DTMF Mode RFC2833) I can login to the
> voicemailsystem, but if I hit # during a call, to transfer it, I get
> warnings and the transfer doesn't work. Seemes like this is a bug or bad
> implementation in the phone?
>
> My setup:
> Asterisk (NT) is connected via 1xE1 (Sangoma A104) to Alcatel 4400 PBX
> (TE). Asterisk gets clock from Alcatel. Do you need that to provide help
> or are you interested how I did it?
>
> chris...
>
> MIVOC Systemes SAS schrieb:
> > Hello,
> >
> > Can you  tell about  your configuration ( connection between Alacaltel
4400
> > and Asterisk ?)  : what  hardware ? what config files ?
> >
> > Thank you
> >
> > Minh VO
> > MIVOC Systemes SAS
> > FRANCE
> > - Original Message -
> > From: "Christoph Fürstaller" <[EMAIL PROTECTED]>
> > To: 
> > Sent: Wednesday, March 14, 2007 9:16 AM
> > Subject: [asterisk-users] strange things on call transfer
> >
> >
> > Hi,
> >
> > I'm setting up an Asterisk system which is connected to an Alcatel 4400
> > PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
> > call by hitting the # key, I get this messages and nothing happens on
> > the phone:
> >
> > WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?  An ilbc frame
> > that isn't a multiple of 50 bytes long from RTP (4)?
> >
> > But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I
> > tet this error and the call is hung up, when transfering:
> >
> > Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> > find a codec translation path from ilbc to slin
> > Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to
> > find a codec translation path from ilbc to slin
> > Mar 14 09:04:58 WARNING[30436]: channel.c:2752
> > ast_channel_make_compatible: No path to translate from Zap/31-1(72) to
> > SIP/374-08199e50(1024)
> > Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't
> > make Zap/31-1 and SIP/374-08199e50 compatible
> > Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call:
> > Bridge failed on channels Zap/31-1 and SIP/374-08199e50
> >
> > Attached there is a SIP debug from such a call (with ilbc loaded)
> >
> > I hope someone can help me.
> >
> > chris...
> >>
> - 
--
> > -
> Orange vous informe que cet  e-mail a ete controle par l'anti-virus mail.
> Aucun virus connu a ce jour par nos services n'a ete detecte.
> >>
> >>
>
>
> --
--
> > 
>
>
> <-- SIP read from 172.28.20.4:2051:
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport
> From: "Test User3" ;tag=48rd3g03e1
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: ;flow-id=1
> P-Key-Flags: keys="3"
> User-Agent: snom300/6.2.3
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> > MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Supported: timer, 100rel, replaces, callerid
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 473
> >>
> v=0
> o=root 1775117380 1775117380 IN IP4 172.28.20.4
> s=call
> c=IN IP4 172.28.20.4
> t=0 0
> m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> > inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:2 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=encryption:optional
> a=sendrecv
> >>
> - --- (18 headers 19 lines) ---
> Using INVITE request as basis request -
> > [EMAIL PROTECTED]
> Sending to 172.28.20.4 : 2051 (NAT)
> Reliably Transmitting (no NAT) to 172.28.20.4:2051:
> SIP/2.0 407 Proxy Authentication

Re: [asterisk-users] Playback 5% Too Fast?

[Tim Panton]

On 13 Mar 2007, at 00:32, David Brazier wrote:


Hi All

I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application.  There are clicks  
every few
seconds or more frequently that is audible on the remote end  
(PSTN), but

not on the Asterisk recording of the call.  If I record the remote end
and compare it to the local recording, it appears to be about 5%-7%  
too
fast - i.e. if I synchronise the starts, the remote end finishes  
sooner.

I can find points in the remote recording where parts of the waveform
have been missed out, leading to jumps in the waveform, which  
correspond
to the audible clicks.  These "jumps" seem like dropped packets,  
and I'm

deducing that Asterisk is sending data slightly too fast (i.e. more
frequently than 50x160 sample per second) for the remote end, which  
has

to drop data to keep up.

This is a VoIP-only set up - no Zap hardware.  Thinking this was a
timing issue, I have installed Zaptel to get ztdummy, which is loaded
OK, but that hasn't made any difference.  I have tried it with  
different

VoIP providers and observed the same problem.

Behaviour has persisted from 1.2 to 1.4 and now 1.4.1.  CentOS 4.4
(2.6.9 kernel), Dell 1950.

Any ideas how to progress?  Is this a timing issue or am I wide of the
mark?

Thanks for any help

David


It would be interesting to see an ethereal trace of the packets going
to your PRI gateway. Ideally the packet capture would be done by
a separate system, so that the clock of your Dell won't also be the
'reference' clock.

Do you run NTP on that system. If you do, take a look at the
skew over a day or so and see if the Dell is running fast.

It might be worth investing in a low end digium card
just to generate a clock that is independent of your CPU
clock.

Tim Panton

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



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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[nivlekch]

nice job moises, the hardwork you and steve put into chan_unicall is 
remarkable.


with a little editing and tweaking, i was able to make
the port to 1.4 here in the philippines without any problems.  some part 
of libmfcr2 has to be changed for proper/better ANI exchage with 
PLDT(telco). looking good so far, better than the experience in 1.2, 
i'll post any update soon.


anybody interfacing with PLDT interested, email me offline.

[EMAIL PROTECTED] wrote:

Im glad to let you know that finally I invested some time to make work
Unicall in Asterisk 1.4, I must say not much testing could be done
since I have no hardware available ( cards, servers ), however a
friend was able to test it with a couple of calls with success, I need
you to test this and report some feedback.

The sources are available in:

http://moy.ivsol.net/unicall/soft-switch/r1b1/

Kind Regards

Moises Silva



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[asterisk-users] What happend to voip-info?

[Nir Simionovich]

Anyone has an idea what happend to voip-info? it stopped working about 24
hours ago.
 
Nir S
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[asterisk-users] Earliest dial tone, after boot up.

[joe acquisto]

New system install.

At what point, in bootup, should I be able to get a dial tone on the phone 
ports on a tdm400p?  There are two fxo and two fxs ports.  I know which to plug 
into .

At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not 
get a dial tone on the phone ports.  I thought as long as zapata.conf is 
correct, the board should be "functional." at that point.

joe a.

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Re: RE: [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking

[Benedikt Franz]

Yep, I have added these lines to the sip.conf (but I see that I now have the 
lines to allow the video codecs in both the sip.conf and in the 
extionsions.conf. I suppose that is not right?):

; Added video support
videosupport=yes
allow=h261
allow=h263
allow=h263p


BTW: How do I properly post a message on this board? If I send an email to 
asterisk-users@lists.digium.com, only about one out of five attempts succeed, 
and the message actually appears in the list.

 Original-Nachricht 
Datum: Wed, 14 Mar 2007 11:32:22 +0300
Von: "Biju" <[EMAIL PROTECTED]>
An: "\'Asterisk Users Mailing List - Non-Commercial Discussion\'" 

CC: 
Betreff: RE: [asterisk-users] Re: Asterisknow with video and X-Lite not 
quiteworking

> Have you added this line in your sip.onf ?
> 
> videosupport=yes 
> 
> Biju
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Benedikt
> Franz
> Sent: Wednesday, March 14, 2007 10:14 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Asterisknow with video and X-Lite not
> quiteworking
> 
> Hi Dave,
> 
> yes, Audio is fine, but no video. And as far as I can see, X-Lite (running
> 3.0 build 34025 here, all Clients have exactly the same version) supports
> only the h.263 and h.263p video codecs. But I am not quite sure if I
> enabled
> these codecs properly. For *now, I have put the allow-lines into the
> users.conf, for instance, heres my setup (I cencored out email and
> secret):
> 
> [6510]
> fullname = Benedikt Franz
> secret = ...
> email = ...
> cid_number = 6510
> zapchan =
> context = numberplan-custom-1
> hasvoicemail = yes
> hasdirectory = yes
> hassip = yes
> hasiax = yes
> hasmanager = yes
> callwaiting = yes
> threewaycalling = yes
> mailbox = 6510
> hasagent = no
> group =
> host = dynamic
> registersip = yes
> registeriax = yes
> allow = h263
> allow = h263p
> canreinvite = yes
> 
> 
> I am not sure if that is a NAT problem, since all users are either on the
> local area network, or connected through VPN (I have not tested video with
> those yet, though), however, I will try that.
> --
> Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen
> downloaden: http://www.gmx.net/de/go/browser
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Re: [asterisk-users] asterisk on mini-itx

[Gordon Henderson]

On Sun, 11 Mar 2007, Ira wrote:


At 01:36 AM 3/11/2007, you wrote:
My servers don't run anything more than they need to and don't have 
packages loaded that they don't need. I could rant on all day about the 
bloat I see in modern RH/Fedora/SuSe, even my favourite Debian systems, but 
this isn't the place ...


I'd love to have my box running that little, but how do I figure out what's 
not needed and how to get rid of it?  One of my frustrations with the Linux 
world is the apparent assumption of people that their target audience already 
knows what they're talking about.


It's hard - unless you are a grumpy old man like me and have been 
"fiddling" with Unix/Linux for years and have the time & patience to prune 
stuff down.


The push right now for the major distros is to get "Linux on the Desktop" 
in a state as good as (or better!) than Windoze, so all sorts of "stuff" 
is included - printing programs, lots of eye candy, hot-plug drivers, 
sound drivers, and a whole host of other life-support code to keep the GUI 
going.


I'm willing to tolerate a lot loss that that, but newbies, or people 
who've previously been exposed to windows might not be.


You could simply start removing packages until it breaks :)
Or do a 'ps ax' and see what's actually running - work out if you need it, 
and remove it. Look in /etc/init.d and so on.


I create my flash boot systems the other awy round - I put on them just 
what I need - I am lazy though, so I have a bare-bones Debian box which is 
my development box. No X windows runs on it, I just ssh in from my main 
workstation. I then copy the whole of /bin /sbin and /lib to the flash 
device, then copy selected programs from /usr/bin. I've also worked out 
exactly which libraries I need from /usr/lib (using ldd on all the 
binaries!) remove any perl scripts in the process, make sure all 
executable files are stipped of their symbol tables and so on. This took 
time, and it's still a "work in progress"...


So I create my main flash drive, then I create a tar-file which contains 
/etc/asterisk, /var/www/bin and a few other selected files & folders. The 
system boots off flash, and early on gets this tar file off a separate 
partition on the flash device and installs it. This way I can update the 
system in 3 phases - kernel, (/boot/bzImage), main system (initrd.gz) or 
just the "software"...


Anyway, this is way off toppic :)

Gordon
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Re: [asterisk-users] Earliest dial tone, after boot up.

[Tzafrir Cohen]

On Wed, Mar 14, 2007 at 07:53:58AM -0400, joe acquisto wrote:
> New system install.
> 
> At what point, in bootup, should I be able to get a dial tone on the 
> phone ports on a tdm400p?  There are two fxo and two fxs ports.  I 
> know which to plug into .

Only after:

* The card's driver has loaded and initialized the card's module.
* It has been configured (ztcfg)
* Asterisk has started

> 
> At boot up, as soon as wctdm is loaded, all the ports "go green, yet I 
> do not get a dial tone on the phone ports.  I thought as long as 
> zapata.conf is correct, the board should be "functional." at that 
> point.

It is Asterisk that actually gives the dialtone. So it is not enough to
have /etc/asterisk/zapata.conf properly configured. You also need
asterisk running.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Single sign on PC + phone?

[Jonathan k. Creasy]

This is an interesting idea, did you come up with anything? 

Are your users logging into an AD domain? A script to interact with the 
Asterisk server could be run after login which adds an extension mapping the 
user to the phone. One set of extensions for the users (which is published) and 
another set of "real" extensions for the phones and when a user extension is 
dialed it rings the phone extension. 

-Jonathan

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick
> Sent: Monday, March 12, 2007 8:33 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Single sign on PC + phone?
> 
> Hi all,
> 
> Does anyone have any experience with creating a Single sign on (SSO)
> concept where if someone logs in on their PC the phone next to that PC
> is also automatically assigned to that user?
> 
> TIA,
> Patrick
> 
> ___
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> No virus found in this incoming message.
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> Version: 7.5.446 / Virus Database: 268.18.9/719 - Release Date: 3/12/2007
> 8:41 AM
> 

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RE: [asterisk-users] What happend to voip-info?

[Steve Totaro]

I have not been able to get to it for a few days.  I offered to mirror
it several times when it was up and down a few years ago and was
declined.  So much good info, bits and pieces that have saved me over
and over.  Let's hope it comes back up.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: Wednesday, March 14, 2007 6:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What happend to voip-info?

 

Anyone has an idea what happend to voip-info? it stopped working about
24 hours ago.

 

Nir S

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RE: [asterisk-users] What happend to voip-info?

[Jonathan k. Creasy]

I would be willing to mirror it also….

 

   _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 14, 2007 9:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What happend to voip-info?

 

I have not been able to get to it for a few days.  I offered to mirror it 
several times when it was up and down a few years ago and was declined.  So 
much good info, bits and pieces that have saved me over and over.  Let’s hope 
it comes back up.

Thanks,
Steve Totaro
HYPERLINK "http://www.asteriskhelpdesk.com"http://www.asteriskhelpdesk.com
KB3OPB
  

   _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich
Sent: Wednesday, March 14, 2007 6:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What happend to voip-info?

 

Anyone has an idea what happend to voip-info? it stopped working about 24 hours 
ago.

 

Nir S


--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 268.18.10/720 - Release Date: 3/12/2007 7:19 
PM



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Checked by AVG Free Edition.
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Re: [asterisk-users] Earliest dial tone, after boot up.

[joe a.]

Thanks.  Very reassuring.   It really must be too early. 

joe a.

Tzafrir Cohen<[EMAIL PROTECTED]> Wrote: 3/14/2007 8:23 AM:
> On Wed, Mar 14, 2007 at 07:53:58AM -0400, joe acquisto wrote:
>> New system install.
>> 
>> At what point, in bootup, should I be able to get a dial tone on the 
>> phone ports on a tdm400p?  There are two fxo and two fxs ports.  I 
>> know which to plug into .
> 
> Only after:
> 
> * The card's driver has loaded and initialized the card's module.
> * It has been configured (ztcfg)
> * Asterisk has started
> 
>> 
>> At boot up, as soon as wctdm is loaded, all the ports "go green, yet I 
>> do not get a dial tone on the phone ports.  I thought as long as 
>> zapata.conf is correct, the board should be "functional." at that 
>> point.
> 
> It is Asterisk that actually gives the dialtone. So it is not enough to
> have /etc/asterisk/zapata.conf properly configured. You also need
> asterisk running.
> 
> -- 
>Tzafrir Cohen   
> icq#16849755jabber:[EMAIL PROTECTED] 
> +972-50-7952406   mailto:[EMAIL PROTECTED]   
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir 
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RE: [asterisk-users] T1 Integrator Birch

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John Schmerold
> Sent: Wednesday, March 14, 2007 1:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] T1 Integrator Birch
> 
> I'm thinking about replacing my Birch T1 integrator with an Asterisk
box.
> 
> The Integrator has 12 voice & 768k data, so the Asterisk box would
> become a router & PBX.
> 
> Has anyone done anything similar, what experiences have you seen &/or
> read about.
> 
> TIA


I tried this with a quad port Digium card a couple of years ago.  It was
not very well documented and I am pretty sure it was not a "supported"
configuration.  It took a lot of digging around to figure out how and
what to enable for the compiling a custom kernel to support the data
mode. 

It may be much simpler now (hope so anyways).  Sangoma seems like it may
be easier to set up, just based on the options in the Sangoma
installation script.

I have no recent experience with it, so please, if you get it working,
post details of what you did, the results and what resources helped you.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB

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[asterisk-users] What is the best phone to get when using a headset?

[Gareth Blades]

We currently have the Grandstream GXP-2000 phones which generally work
very well except that we cannot get find a headset which works reliably
with them. Either the sound quality is poor or the other party has
difficulty in hearing us.

We therefore want to get a couple of different phones and headsets for
our customer service people.
What would you recommend as a good phone and headset combination?

Thanks

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[asterisk-users] Autoprovisioning ST2030S

[Christoph Fürstaller]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

Anyone tried autoprovisioning for a Thomson ST2030S phone?

I did so and are quiet happy with it except for one thing: DTMF settings

Whad parameter in the provisioning files do I have to set to transmit
DTMF via SIP INFO? I accomplished Inband and Outband (RFC2833) but I
cant get it to set to SIP INFO.

Parameter is the first one on the advanced_advanced.html webpage.

Anyone knows how I can do that?

christoph
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.3 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF9/YbR0exH8dhr/YRAsgcAKCyPHN2j/XWKCAKlPxroexNhYbizwCfUCZ0
3roYvY8mL5zIqPtOPPNpmJM=
=EKrD
-END PGP SIGNATURE-
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RE: [asterisk-users] What is the best phone to get when using a headset?

[Cory Andrews]

I like any of the Polycom Soundpoint IP Series phones (IP301, 430, 501,
601, 650) paired with a VXI Tuffset 37 with Quick Disconnect Cord.  IMO
VXI makes the best headsets out there for the money, and they are quite
inexpensive.

You can find more info on VXI at www.vxicorp.com


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
Blades
Sent: Wednesday, March 14, 2007 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What is the best phone to get when using a
headset?

We currently have the Grandstream GXP-2000 phones which generally work
very well except that we cannot get find a headset which works reliably
with them. Either the sound quality is poor or the other party has
difficulty in hearing us.

We therefore want to get a couple of different phones and headsets for
our customer service people.
What would you recommend as a good phone and headset combination?

Thanks

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RE: [asterisk-users] Nomination for Coolest App in 2007

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brad Templeton
> Sent: Monday, March 12, 2007 8:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Nomination for Coolest App in 2007
> 
> On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote:
> > Mine goes to chan_bluetooth.  Somewhat of a pain getting it going
but I
> > am totally floored with how cool it is!
> >
> > Thanks,
> > Steve Totaro
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> My, that is a cool app.  I look forward to running it when it's a bit
more
> stable.   While the outgoing call ability seems of limited use since
> cell call quality is not that exciting to even unlimited night and
> weekend minuets are probably not too attractive compared to 1
cent/minute
> SIP terminations, there are a number of interesting possibilities:
> 
> a) If your target has an unlimited calls to other
> customers/family/etc.
> plan, you would want to call them this way to save minutes.
> b) Handy on some carriers for checking cell voice mail.  (I have
> found that with many US carriers, however, you can call
> your cell phone with CID set to your cell number, and it goes
> directly to voice mail  Make sure you have a password!)
> c) Incoming calls, obviously handy.
> d) During daytime, program to receive incoming calls and say,
> "I am at my desk.  Please call me at xxx- or press 1 to
> have me call you back at " so you get
> better quality and don't bill cell minutes.  In the evening,
> assuming unlimited weekends, you might forward directly.
> 
> Can it send and receive SMS via bluetooth too?
> 
> 
> I also like a lot the talk of coming softphones with bluetooth
> headset support.   This would allow you to use your bluetooth
> headset as an extension on your Asterisk pbx.   I happen to have
> a bluetooth headset that plugs into my hard phone -- I wish more
> hardphones supported them natively -- and that's handy.  This could
> be just as good.   To really get it right you would want some
> speech recognition so you could place calls from the bluetooth
> headset by saying names and digits, as many cell phones can already
> do.
> 
> Of course, a linux softphone could reside right on the asterisk box.
> You could multi-dial your bluetooth headset and your hard phones and
> answer where you like.

I was part of an onsite USAID project to evaluate and help rebuild the
infrastructure in war torn West Africa (Sierra Leone, Liberia, Guinea).
A great experience to truly realize what down and out is, but more
importantly, see what real hope and the excitement that a new democracy
is.

Interestingly, when war broke out in these areas, land lines were cut.
Probably partially to prevent communication but I was told it was
actually more the locals that used the copper in the lines to make pots
and pans.

Everything was prepaid cellular.  GSM gateways were the only way to
connect a PBX and they were quite expensive.  With this app, a smaller
NGO with not little funding could setup a nice PBX a few cell phones
doing triple duty as extensions, FXO ports, and an SMS gateway.
Everything is pre-paid there since there is no such thing as credit. 

Another interesting (from an American's perspective anyways) is that
inbound calls on cell phones are free.  Even if you buy a SIM with a
little pre-paid time and use up the time, you can still receive inbound
calls for free for a couple months.  

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB


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RE: [asterisk-users] What is the best phone to get when using a headset?

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Gareth Blades
> Sent: Wednesday, March 14, 2007 7:54 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] What is the best phone to get when using a
> headset?
> 
> We currently have the Grandstream GXP-2000 phones which generally work
> very well except that we cannot get find a headset which works
reliably
> with them. Either the sound quality is poor or the other party has
> difficulty in hearing us.
> 
> We therefore want to get a couple of different phones and headsets for
> our customer service people.
> What would you recommend as a good phone and headset combination?
> 
> Thanks

It depends how many you mean by a couple.  If it is really just a
couple, you could just get an FXS card and use plain old analog headsets
on a single pair.  That is probably what I would do, especially if your
customer support staff beats up on equipment.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB


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[asterisk-users] Manager connection problems

[Jordan Novak]

 
I am wondering how many and how often manager connections can be setup
and torn down reasonably.

 

here is the scenerio...

I have 10 to 20 agents on two queues

one with priority over the other

I changed this the day before

I also implemented a php program that runs every 8 seconds on an
automatic refresh

It establishes a connection to asterisk and runs a mysql query to update
the database

It runs two commands, Agents and QueueStatus

During the weekend the system dropped all the calls and logged out all
the agents and came back on its own.

I am running asterisk without "safe" or any other scripts

I run a Te410p with forthy eight channels to a pbx

I can't find one error in any logs, I am unsure where to look in
asterisk though.

 

Basically I have a logon and off twice every 10 seconds, is this the
problem! I implemented the priorities within queues.conf, does that
cause any problems.

I would appreciate any feedback!!!
 
 
Jordan Novak
Senior Telecommunications Engineer
Logistics Health Inc.
1319 Saint Andrews Street 
La Crosse,WI 54603
(608) 783-7560 x1299
1-800-666-2833 x1299
 
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Re: [asterisk-users] Playback 5% Too Fast?

[Cosmin Prund]

I've had similar behavior on my own IVR. I moved my sound files to a ram 
disk and all pops and ticks stopped!


David Brazier wrote:

Hi All

I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application.  There are clicks every few
seconds or more frequently that is audible on the remote end (PSTN), but
not on the Asterisk recording of the call.  If I record the remote end
and compare it to the local recording, it appears to be about 5%-7% too
fast - i.e. if I synchronise the starts, the remote end finishes sooner.
I can find points in the remote recording where parts of the waveform
have been missed out, leading to jumps in the waveform, which correspond
to the audible clicks.  These "jumps" seem like dropped packets, and I'm
deducing that Asterisk is sending data slightly too fast (i.e. more
frequently than 50x160 sample per second) for the remote end, which has
to drop data to keep up.  


This is a VoIP-only set up - no Zap hardware.  Thinking this was a
timing issue, I have installed Zaptel to get ztdummy, which is loaded
OK, but that hasn't made any difference.  I have tried it with different
VoIP providers and observed the same problem.

Behaviour has persisted from 1.2 to 1.4 and now 1.4.1.  CentOS 4.4
(2.6.9 kernel), Dell 1950.

Any ideas how to progress?  Is this a timing issue or am I wide of the
mark?

Thanks for any help

David
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RE: [asterisk-users] Manager connection problems

[Steve Totaro]

What version of Asterisk?  I had a lot of problems when hitting the
manager interface with any real volume with 1.2.x, it is my
understanding that 1.4.x has a newer revamped AMI that should be more
robust.  I haven't tried it yet so I cannot confirm.

 

Another major thing you should check is your queueprio setting.  In one
installation I managed, this setting made Asterisk (and Linux) become
completely unresponsive.  There is a warning about it on voip-info (if
anyone could get to it).  Removing queueprio fixed the problem in this
case.  

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
Novak
Sent: Wednesday, March 14, 2007 8:41 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Manager connection problems

 

 

I am wondering how many and how often manager connections can be setup
and torn down reasonably.

 

here is the scenerio...

I have 10 to 20 agents on two queues

one with priority over the other

I changed this the day before

I also implemented a php program that runs every 8 seconds on an
automatic refresh

It establishes a connection to asterisk and runs a mysql query to update
the database

It runs two commands, Agents and QueueStatus

During the weekend the system dropped all the calls and logged out all
the agents and came back on its own.

I am running asterisk without "safe" or any other scripts

I run a Te410p with forthy eight channels to a pbx

I can't find one error in any logs, I am unsure where to look in
asterisk though.

 

Basically I have a logon and off twice every 10 seconds, is this the
problem! I implemented the priorities within queues.conf, does that
cause any problems.

I would appreciate any feedback!!!

 

 

Jordan Novak

Senior Telecommunications Engineer

Logistics Health Inc.

1319 Saint Andrews Street 

La Crosse,WI 54603

(608) 783-7560 x1299

1-800-666-2833 x1299

 

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[asterisk-users] Re: asterisk on mini-itx

[Tomislav Parcina]

Gordon Henderson wrote:
I've built several systems based on this motherboard (the 1GHz fanless 
one) Compressed codecs are fine - as long as you aren't transcoding ;-) 
I figured I could push 30 non transcoded calls through one, but I've 
never had the ability to fully test it out. The max. I had going on one 
system was 20 calls.


What was CPU usage during this 20 calls?


--
Tomislav Parcina
[EMAIL PROTECTED]

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RE: [asterisk-users] Manager connection problems

[Steve Totaro]

Correction, it was queue weight that caused the crashes not queue_prio.

 

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Wednesday, March 14, 2007 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Manager connection problems

 

What version of Asterisk?  I had a lot of problems when hitting the
manager interface with any real volume with 1.2.x, it is my
understanding that 1.4.x has a newer revamped AMI that should be more
robust.  I haven't tried it yet so I cannot confirm.

 

Another major thing you should check is your queueprio setting.  In one
installation I managed, this setting made Asterisk (and Linux) become
completely unresponsive.  There is a warning about it on voip-info (if
anyone could get to it).  Removing queueprio fixed the problem in this
case.  

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
Novak
Sent: Wednesday, March 14, 2007 8:41 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Manager connection problems

 

 

I am wondering how many and how often manager connections can be setup
and torn down reasonably.

 

here is the scenerio...

I have 10 to 20 agents on two queues

one with priority over the other

I changed this the day before

I also implemented a php program that runs every 8 seconds on an
automatic refresh

It establishes a connection to asterisk and runs a mysql query to update
the database

It runs two commands, Agents and QueueStatus

During the weekend the system dropped all the calls and logged out all
the agents and came back on its own.

I am running asterisk without "safe" or any other scripts

I run a Te410p with forthy eight channels to a pbx

I can't find one error in any logs, I am unsure where to look in
asterisk though.

 

Basically I have a logon and off twice every 10 seconds, is this the
problem! I implemented the priorities within queues.conf, does that
cause any problems.

I would appreciate any feedback!!!

 

 

Jordan Novak

Senior Telecommunications Engineer

Logistics Health Inc.

1319 Saint Andrews Street 

La Crosse,WI 54603

(608) 783-7560 x1299

1-800-666-2833 x1299

 

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[asterisk-users] Zaptel version for asterisk 1.2.16

[Wilson Pickett]

I'm used to seeing the same versioning (maybe I've been gone too long)

Is zaptel 1.2.15 the right one for asterisk 1.2.16 ?
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Re: [asterisk-users] How many outgoing phone line/voip account do I need?

[Lee Jenkins]

Kurt Kuo wrote:

Hi list,
I have an application which has to automatically dial and send out a 
voice message to 50 different phone numbers at the same time. Does it 
mean that I need to sign up 50 phone lines or voip accounts in order to 
achieve this purpose? Is there a provider(voip prefer) who offer a 
special account which is able to handle multiple calls simultaneously?

Thanks in advance.



Hi,

Just curious.  When you say "at the same time", do you mean they MUST 
all be made at the same time or do you mean make all 50 calls in a 
relatively short interval?  That would make a big difference, I think.


I ask because you say that you want to "send out a voice message" and it 
sounds like you just want to broadcast a message to 50 persons, not 
necessarily dialing all 50 at the same time.


If you do not need to make all 50 calls at the same time, you can break 
them up into 5,10,20 call attempts at a time until all recipients have 
been reached or they have been tried the maximum number of times...


--

Warm Regards,

Lee


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[asterisk-users] what happened to asterisk wiki???

[Rizwan Hisham]

Hi
im trying access the www.voip-info.org website since yesterday but i cant
open it. google search diaplay correct search results but it doesnt open
when i click the link. it displays a message about tcp error which says
-->"There was a problem communicating with the server". I dont know what the
problem is. I just want to ask whether their server is down or not and is
everybody having the same problem?

--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] What is the best phone to get when using a headset?

[Bruce Reeves]

Cory,

Are the Polycom phones able to detect that the headset is off-hook? We
have had problems with 2 different brands of headsets working fine but
the phones seem unaware of the headset's hook state.

On 3/14/07, Cory Andrews <[EMAIL PROTECTED]> wrote:

I like any of the Polycom Soundpoint IP Series phones (IP301, 430, 501,
601, 650) paired with a VXI Tuffset 37 with Quick Disconnect Cord.  IMO
VXI makes the best headsets out there for the money, and they are quite
inexpensive.

You can find more info on VXI at www.vxicorp.com


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
Blades
Sent: Wednesday, March 14, 2007 8:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What is the best phone to get when using a
headset?

We currently have the Grandstream GXP-2000 phones which generally work
very well except that we cannot get find a headset which works reliably
with them. Either the sound quality is poor or the other party has
difficulty in hearing us.

We therefore want to get a couple of different phones and headsets for
our customer service people.
What would you recommend as a good phone and headset combination?

Thanks

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--
Bruce
Nortex Networks
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Re: [asterisk-users] what happened to asterisk wiki???

[Lee Jenkins]

Rizwan Hisham wrote:

Hi
im trying access the www.voip-info.org  
website since yesterday but i cant open it. google search diaplay 
correct search results but it doesnt open when i click the link. it 
displays a message about tcp error which says -->"There was a problem 
communicating with the server". I dont know what the problem is. I just 
want to ask whether their server is down or not and is everybody having 
the same problem?


--


Richard,

The site seems to be down right now.  Interesting how we take it for 
granted, no?


There was a thread earlier where someone asked about the wiki as well.

I hope its not down for long...

Warm Regards,

Lee


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RE: [asterisk-users] what happened to asterisk wiki???

[Steve Totaro]

It is down.  Two options, search in Google, click on cached and then
click on cached text only at the top.  

 

This may also be helpful
http://web.archive.org/web/*/http://www.voip-info.org

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, March 14, 2007 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] what happened to asterisk wiki???

 

Hi
im trying access the www.voip-info.org website since yesterday but i
cant open it. google search diaplay correct search results but it doesnt
open when i click the link. it displays a message about tcp error which
says -->"There was a problem communicating with the server". I dont know
what the problem is. I just want to ask whether their server is down or
not and is everybody having the same problem? 

-- 
Regards
Rizwan Hisham
Software Engineer 

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Re: [asterisk-users] Asterisknow with video and X-Lite not quite working

[Pari Nannapaneni]

I request every one to post AsteriskNOW specific questions on the
asteriskNOW forums - http://forums.digium.com/

I will talk to our administrator to see if i can get a seperate mailing
list created for AsteriskNOW.

thanks
Pari


Pari Nannapaneni
GUI Developer
Digium Inc.



Benedikt Franz wrote:

Hello everyone,

I have previously asked this question on the asterisk-video list, but I 
got directed here.


I have a setup consisting of asterisknow beta4 (not sure if that is 
crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the 
local network. My computer has a USB-Camera installed, and now I would 
like to do some video calling with it, at least, so that the other user 
can see me.


When I make a call and then click 'Start' (sending video) in the X-Lite 
client, nothing seems to happen on the other side, but here it says that 
a video transmission has begun. According to 'sip show codecs', both the 
h.263 and h.263p codec are supported, and those are also set on either 
X-Lite clients. I have enabled 'canreinvite' for both users as well, but 
still the other user can not see me. I can, however, see the cameras 
view on my computer, so that seems all properly set up.


Could anyone help me sort this out?

Thanks.


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Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

[Moises Silva]

nivlekch, nice to hear that :)

I hope more people can test this.

On 3/14/07, nivlekch <[EMAIL PROTECTED]> wrote:

nice job moises, the hardwork you and steve put into chan_unicall is
remarkable.

with a little editing and tweaking, i was able to make
the port to 1.4 here in the philippines without any problems.  some part
of libmfcr2 has to be changed for proper/better ANI exchage with
PLDT(telco). looking good so far, better than the experience in 1.2,
i'll post any update soon.

anybody interfacing with PLDT interested, email me offline.

[EMAIL PROTECTED] wrote:
> Im glad to let you know that finally I invested some time to make work
> Unicall in Asterisk 1.4, I must say not much testing could be done
> since I have no hardware available ( cards, servers ), however a
> friend was able to test it with a couple of calls with success, I need
> you to test this and report some feedback.
>
> The sources are available in:
>
> http://moy.ivsol.net/unicall/soft-switch/r1b1/
>
> Kind Regards
>
> Moises Silva
>

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Re: [asterisk-users] Asterisknow with video and X-Lite not quite working

[Benedikt Franz]

I do not think that this is a specifically *now related issue, but I would also 
welcome such a mailing list.

Regards

 Original-Nachricht 
Datum: Wed, 14 Mar 2007 10:22:34 -0500
Von: Pari Nannapaneni <[EMAIL PROTECTED]>
An: Asterisk Users Mailing List - Non-Commercial Discussion 

CC: 
Betreff: Re: [asterisk-users] Asterisknow with video and X-Lite not quite   
working

> I request every one to post AsteriskNOW specific questions on the
> asteriskNOW forums - http://forums.digium.com/
> 
> I will talk to our administrator to see if i can get a seperate mailing
> list created for AsteriskNOW.
> 
> thanks
> Pari
> 
> 
> Pari Nannapaneni
> GUI Developer
> Digium Inc.
> 
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[asterisk-users] [G.729] Input Gain

[Victor Mateevitsi]

Hello,

I have recently buyed the g.729 license from the Digium site and have the
following issue:

I have two voip providers (SIP). The first uses g.726 and the second g.729.
The problem is that the input gain from the second provider is a little
lower than the first one. I usually use both, so configuring the input gain
on the IP-Phones is not a solution.

Can I configure the input gain of a context on sip.conf or maybe can I
configure the input gain of g.729 encoded streams. In zapata.conf (or
misdn.conf) I know there is the above option (rxgain and txgain).

Thank you in advance,
Victor
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RE: [asterisk-users] What happend to voip-info?

[Gordon Henderson]

On Wed, 14 Mar 2007, Jonathan k. Creasy wrote:


I would be willing to mirror it also?.


At the risk of sounding like an AOLer, Me Too ... (UK based mirror?)

The site is pingable, so I'd suggest it's either crashed in some awkward 
way and just needs resetting, but you never know...


Gordon





  _

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 14, 2007 9:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What happend to voip-info?



I have not been able to get to it for a few days.  I offered to mirror it 
several times when it was up and down a few years ago and was declined.  So 
much good info, bits and pieces that have saved me over and over.  Let?s hope 
it comes back up.

Thanks,
Steve Totaro
HYPERLINK "http://www.asteriskhelpdesk.com"http://www.asteriskhelpdesk.com
KB3OPB


  _

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich
Sent: Wednesday, March 14, 2007 6:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] What happend to voip-info?



Anyone has an idea what happend to voip-info? it stopped working about 24 hours 
ago.



Nir S


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PM



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[asterisk-users] IVR after hangup

[Benny Amorsen]

I have a rather interesting issue with catching calls which have been
hung up by the callee, but not by the caller. I would like those calls
to return to an IVR, and it almost works:

[incoming]
 exten => 12345678,1,Goto(testIVR,s,1)

[testIVR]
 exten => s,1,Answer
 exten => s,n(play),Background(beep)
 exten => s,n,WaitExten(5)

 exten => 1,1,Dial(SIP/phone)

 exten => t,1,Goto(s,play)

 exten => i,1,Goto(s,play)

 exten => h,1,Playback(01)
 exten => h,n,Wait(1)
 exten => h,n,Goto(testIVR,s,1)


A call comes in through incoming, gets sent off the testIVR, the
caller dials 1. The SIP phone answers, talks, and hangs up. Now the
caller is sent back to the IVR, and hears the beep. It all works
perfectly until the caller dials 1. Then the call is abrubtly cut off,
instead of going through the Dial().


/Benny


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[asterisk-users] Linksys not Ringing

[Jason Walker]

I have 2 linksys SIP phones SPA-942
I have  a dialplan of

exten => 144,1,Wait(1)
exten => 144,2,Dial(Sip/phil,20)
exten => 144,3,Voicemail([EMAIL PROTECTED],u)

The CLI looks like this when I dial 144

-- Executing Wait("IAX2/JASONSERVER-9", "1") in new stack
   -- Executing Dial("IAX2/JASONSERVER-9", "Sip/phil|20") in new stack
   -- Called phil
   -- Nobody picked up in 2 ms
   -- Executing VoiceMail("IAX2/JASONSERVER-9", "[EMAIL PROTECTED]|u") in 
new stack

   -- Playing 'vm-theperson' (language 'en')

It is registered and will make calls but I never get the
  -- SIP/phil is ringing

This happening on my 2 linksys phones only

Jason
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RE: RE: [asterisk-users] Re: Asterisknow with video and X-Lite notquiteworking

[Biju]

Hi,

Codec details you need to put only in the sip.conf 


Biju.V.P

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benedikt Franz
Sent: Wednesday, March 14, 2007 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE: [asterisk-users] Re: Asterisknow with video and X-Lite
notquiteworking

Yep, I have added these lines to the sip.conf (but I see that I now have the
lines to allow the video codecs in both the sip.conf and in the
extionsions.conf. I suppose that is not right?):

; Added video support
videosupport=yes
allow=h261
allow=h263
allow=h263p


BTW: How do I properly post a message on this board? If I send an email to
asterisk-users@lists.digium.com, only about one out of five attempts
succeed, and the message actually appears in the list.

 Original-Nachricht 
Datum: Wed, 14 Mar 2007 11:32:22 +0300
Von: "Biju" <[EMAIL PROTECTED]>
An: "\'Asterisk Users Mailing List - Non-Commercial Discussion\'"

CC: 
Betreff: RE: [asterisk-users] Re: Asterisknow with video and X-Lite not
quiteworking

> Have you added this line in your sip.onf ?
> 
> videosupport=yes
> 
> Biju
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Benedikt 
> Franz
> Sent: Wednesday, March 14, 2007 10:14 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: Asterisknow with video and X-Lite not 
> quiteworking
> 
> Hi Dave,
> 
> yes, Audio is fine, but no video. And as far as I can see, X-Lite 
> (running 3.0 build 34025 here, all Clients have exactly the same 
> version) supports only the h.263 and h.263p video codecs. But I am not 
> quite sure if I enabled these codecs properly. For *now, I have put 
> the allow-lines into the users.conf, for instance, heres my setup (I 
> cencored out email and
> secret):
> 
> [6510]
> fullname = Benedikt Franz
> secret = ...
> email = ...
> cid_number = 6510
> zapchan =
> context = numberplan-custom-1
> hasvoicemail = yes
> hasdirectory = yes
> hassip = yes
> hasiax = yes
> hasmanager = yes
> callwaiting = yes
> threewaycalling = yes
> mailbox = 6510
> hasagent = no
> group =
> host = dynamic
> registersip = yes
> registeriax = yes
> allow = h263
> allow = h263p
> canreinvite = yes
> 
> 
> I am not sure if that is a NAT problem, since all users are either on 
> the local area network, or connected through VPN (I have not tested 
> video with those yet, though), however, I will try that.
> --
> Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen
> downloaden: http://www.gmx.net/de/go/browser 
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RE: [asterisk-users] Zaptel version for asterisk 1.2.16

[Yuan LIU]

From: "Wilson Pickett" <[EMAIL PROTECTED]>
Date: Wed, 14 Mar 2007 15:18:35 +0100

I'm used to seeing the same versioning (maybe I've been gone too long)

Is zaptel 1.2.15 the right one for asterisk 1.2.16 ?


It works.  I've tried some other mixes and they also work.

Yuan Liu


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[asterisk-users] ${EXTEN} is limited to 17 characters under IAX ?

[Oded Arbel]

Hi list. 

We have a problem when dialing over IAX to another Asterisk server:
we've setup an extension named
'f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1' in a specific context (its
large because it needs to be unique). I've read in past discussions on
asterisk-dev list that the extension length is limited to 79 characters
- which I though should be more then enough.

Now were doing a DUNDi lookup on that extension and dialing to it from a
second Asterisk server. The dial address looks like this:
IAX2/dundi-context:[EMAIL PROTECTED]/f19dffb971b93746d73ec46d5f1d4b36c199f48c-g1

The problem is that on the local server, we try to read ${EXTEN} and
parse it (specifically - I want to get at the 'g1' at the end. for this
I use the CUT function):
[mydundictx]
exten => _[0-9a-fA-f_].,1,Set(lastpart=${CUT(EXTEN,,2)})
exten => _[0-9a-fA-f_].,2,Set(firstpart=${CUT(EXTEN,,1)})

and then we get this (in the console):

 -- Accepting AUTHENTICATED call from 192.118.54.135: [...]
-- Executing [EMAIL PROTECTED]:2]
Set("IAX2/192.118.54.135:4569-1", "lastpart=") in new stack
-- Executing [EMAIL PROTECTED]:3]
Set("IAX2/192.118.54.135:4569-1", "firstpart=f19dffb971b93746d") in new
stack

I understand that Asterisk truncates the extension in the display (in
this case - to 17 characters), but I was under the impression that this
is for display only. Apparently this is not the case - the as evidently
at least CUT sees only the first 17 characters ?!?

Then we changed the setup to dial from server to server using SIP
instead of IAX2 - using this method, the entire extension is passed
correctly. Any idea whats going on here ?

We're using Asterisk 1.4.0.

-- 
Oded Arbel
Atelis
[EMAIL PROTECTED]
Tel: +972-54-7340014
::..
After a number of decimal places, nobody gives a damn.

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[asterisk-users] Call center manager for Asterisk (Release 0.3)

[nik600]

Hi

i just want to let you know that is available a new release of ccmanager.

I've added the possibility to import queue_log information in a mysql
database and to generate reports using this information.

The software is in a beta state and provides this functionality:

- users management
- call generation (making a GET or POST request on a certain URL)
- queue management (LOGIN / LOGOUT / QUEUE STATUS)
- pickup a call from a queue even if the user isn't logged in the queue
- outbound call in customizable context
- queue stats import from queue_log
- queue reports creation (using an open xml format)

Please note, i think that the xml definition of a report is very
important, if many people share each other their reports there is the
possibility to build a reports-repository, so the final user can use
many reports and, if the user know what he is doing, he can customize
the reports.

I am looking for people to improve this project, any help would be appreciated.

- developers (php / mysql / postgres / ajax )
- tester
- graphics (div & css)

Here there are some screenshots

https://sourceforge.net/dbimage.php?id=115442
https://sourceforge.net/dbimage.php?id=115440
https://sourceforge.net/dbimage.php?id=114381

And here there is the sourceforge project.

https://sourceforge.net/projects/ccmanager

Thanks, nik
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Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

[Mojo with Horan & Company, LLC]

IIRC, you need an extension named 'callpark' in your extensions.conf 
that calls the ParkAndAnnounce application.


This should get you started:

exten => 
callpark,1,ParkAndAnnounce(PARKED|600|Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]|incoming,s,1)


in the CLI:
Show Application ParkAndAnnounce
for usage info.

The third field, in my case "Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]" 
is the channel to announce the parked call slot to.  In my case, 
extensions beginning with 1xx are the phones themselves, and extensions 
4xx are the same phones but will make them auto-answer (like paging). 
You might have a better way to do this because this is a little cumbersome.


hope it helps!

Mojo

Stephen Bosch wrote:

Hi:

I want to make parking calls easier for my hard-working users. Is there
a way to make the Polycom call park feature work with Asterisk?

In case it just works out of the box, I haven't tried it yet; but the
"call park" feature isn't enabled on the Polycom phones by default.

-Stephen-
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[asterisk-users] IAX2 - Congestion

[Mario Mayerle Filho]

Hy all!

Your Asterisk server is return this log :

*CLI> -- Executing Dial("Khomp/B0C0", "IAX2/*.*.*.*/9834|30|r") in new stack
-- Called *.*.*.*/9834
Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting 
call due to slow response
-- IAX2/*.*.*.*:4569-1 is circuit-busy
-- Hungup 'IAX2/*.*.*.*:4569-1'
  == Everyone is busy/congested at this time (1:0/1/0)


When someone use the IAX2 trunk.
Can anyone helpme?

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[asterisk-users] Which SIP method/option to display a short text message ?

[Olivier]

Hi,

Using SIP methods and options, is there any way for a callee to send the
caller a short text message when the call is establishing ?

Scenario is :

Alice and Bob's SIP phones are registered to an Asterisk server.
Alice calls Bob : an INVITE message is sent to Bob's phone
Bob is replying : a 200 OK message is sent back to Alice with a short text
included ("Welcome to BoB Corporation")
Alice's phone acknowledges (ACK) and displays "Welcome to BoB Corporation"
somewhere beside the dialed digits (Bob's number).

Regards
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[asterisk-users] Sped up recordings with 1.4.1

[Joshua Thompson]

Normally I manage to figure stuff out on my own, but this one is driving me
absolutely bonkers...

I'm working with an Asterisk box at work, running 1.4.1 on a Core2 Duo
machine with kernel 2.6.19.1. There's a quad T1 card (older tor2) in the
machine, though this problem happens if I just use ztdummy too. The box
isn't actually on the PSTN yet so I'm testing with Ekiga, using uncompressed
ulaw audio.

The issue I'm seeing is that  any time I try to record audio to make prompts
the result comes back sounding sped up, but the pitch is the same. It's
sounds like it has dropped every other audio packet. At first I thought it
was the Dictate app so I did a quick extension hack with Record but it does
the same thing. It's definitely a recording issue because if I transfer the
recorded files to my workstation and play them back manually they sound the
same.

What's REALLY got me confused though is that if I run an echo test, it works
beautifully...crystal clear audio with no discernible delays or distortions.
I am taking that to mean my network connection between workstation and PBX
is ok and that Ekiga is transmitting audio ok.

I've quadruple checked everything I can think of, short of replacing the
entire box. The results from zttest are rock solid at 99.987% both with just
ztdummy and with the card, and the machine doesn't exhibit any symptoms of
hardware issues. I've spent a good half a day drudging through google and
the Digium bug reporting system and can't find any reference to this
problem.

Any ideas?
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Re: [asterisk-users] Zaptel version for asterisk 1.2.16

[John Novack]

Wilson Pickett wrote:

I'm used to seeing the same versioning (maybe I've been gone too long)

Is zaptel 1.2.15 the right one for asterisk 1.2.16 ?
In Digium's infinite wisdom, they have seen fir to have version numbers 
no longer match, and also NOT provide any sort of map to give the rest 
of us a clue as to what goes with what.


Probably to discourage wider use of the product?

John Novack


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[asterisk-users] IAX2 and Faxing

[Davis Sylvester III]

Can someone tell me how to get faxing to work with an IAX2 remote client?

I have two customers both are connected via Digium's ATA's S101I both 
can place and receive calls without any problems.  However when they 
connect there line to a fax machine it never connects to the remote fax 
machine.  They place a normal POTS line and all faxes fine.


Any light is appreciated.

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Re: [asterisk-users] IAX2 and Faxing

[Lee Howard]

Davis Sylvester III wrote:


Can someone tell me how to get faxing to work with an IAX2 remote client?



Wish upon a falling star, throw a penny into a fountain, find a 
four-leaf clover, and pat your head and rub your stomach all at the same 
time.  :-)


http://hylafax.sourceforge.net/docs/fax-over-voip.pdf

For the most part it's not going to happen, at least reliably.

I have two customers both are connected via Digium's ATA's S101I both 
can place and receive calls without any problems.  However when they 
connect there line to a fax machine it never connects to the remote 
fax machine.  They place a normal POTS line and all faxes fine. 



Read the PDF doc, and maybe you'll understand why voice calls are fine 
but not fax... why POTS faxing is fine but not over VoIP.


Lee.
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[asterisk-users] RE: what happened to asterisk wiki???

[JR Richardson]

A friend of mine was on the site yesterday, late morning, when he
refreshed his screen, a banner came across the web page "VOIP SUCKS"
and then the site was no longer available.  I'm pretty sure the site
was compromised by some hacker trying to prove a point or make a
statement.  Not to throw stink on anyone or group, but maybe it was
someone from a competing open source VoIP project or one of the Big
Iron VoIP System Manufacturers.  Probably just some cracker with too
much time on their hands.  I feel like someone shot my dog, please get
the site back up as soon as possible.

Thanks.

JR

--
JR Richardson
Engineering for the Masses
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[asterisk-users] Packetization Rate

[Matt]

To my knowledge, Asterisk's packetization rate is hard coded at 30ms.  If I
wanted to, where in the code could I go to change it to 20ms.   Is there
anything bad that might happen if I change it (asterisk related)?
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Re: [asterisk-users] IAX2 and Faxing

[Davis Sylvester III]

Lee Howard wrote:

Davis Sylvester III wrote:

Can someone tell me how to get faxing to work with an IAX2 remote 
client?



Wish upon a falling star, throw a penny into a fountain, find a 
four-leaf clover, and pat your head and rub your stomach all at the 
same time.  :-)


http://hylafax.sourceforge.net/docs/fax-over-voip.pdf

For the most part it's not going to happen, at least reliably.

I have two customers both are connected via Digium's ATA's S101I both 
can place and receive calls without any problems.  However when they 
connect there line to a fax machine it never connects to the remote 
fax machine.  They place a normal POTS line and all faxes fine. 



Read the PDF doc, and maybe you'll understand why voice calls are fine 
but not fax... why POTS faxing is fine but not over VoIP.


Lee.
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Thanks for the quick response.  But how do other VoIP Service Providers 
like vonage provide fax capabilities?


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Re: [asterisk-users] IAX2 and Faxing

[Lee Howard]

Davis Sylvester III wrote:

Thanks for the quick response.  But how do other VoIP Service 
Providers like vonage provide fax capabilities?



Inevitably what I say here is going to be countered from some Vonage 
customer saying, "You're wrong!  It always has worked for me."  Vonage 
even advertises fax service.  I think that you have to pay extra for it, 
but they claim to provide it.  However, I've also known many unhappy 
Vonage customers who couldn't get a fax through to save their life.  
Read the doc I suggested, and hopefully you'll understand why their 
experience can simply be a luck of the draw.


Now, I'm not a Vonage expert... so I don't know exactly what they're 
doing.  So I don't know what is happening differently for those happy 
Vonage customers.  But let me tell you some things that could be 
happening to mitigate their situation.


1.  The distance or number of hops between the user and their 
Vonage-controlled gateway could be very, very short... which would allow 
Vonage a lot better chance of controlling the problematic fax-over-VoIP 
factors.


2. Vonage could actually be using T.38 (or perhaps T.37) in their ATAs 
and gateways.  I think that this is unlikely, but possible.


3. Undoubtedly the Vonage ATA uses QoS, and if everyone in the path 
between the user and the Vonage gateway honors that QoS to some degree 
or another, then maybe that's helping and making the difference.


4. Vonage could be employing some fax-specific kind of jitterbuffer on 
both ends of the VoIP connection.  This would help control the 
jitter-related problems... but again, it wouldn't solve the problem of 
audio corruption happening.


5. Vonage could be acting as a store-and-forward fax relay, employing an 
extremely tolerant set of fax protocols to mitigate the jitter issues.  
I think that this is unlikely, but possible.


6. They could be using some combination of the above or something else 
that I've not thought of right now.


As for my guess... I'd suspect that they're using #3 and #4 together, 
and that #1 comes into play more frequently than we would normally think.


Lee.
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Re: [asterisk-users] RE: what happened to asterisk wiki???

[Matt Riddell (NZ)]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

JR Richardson wrote:
> A friend of mine was on the site yesterday, late morning, when he
> refreshed his screen, a banner came across the web page "VOIP SUCKS"
> and then the site was no longer available.  I'm pretty sure the site
> was compromised by some hacker trying to prove a point or make a
> statement.  Not to throw stink on anyone or group, but maybe it was
> someone from a competing open source VoIP project or one of the Big
> Iron VoIP System Manufacturers.  Probably just some cracker with too
> much time on their hands.  I feel like someone shot my dog, please get
> the site back up as soon as possible.

There was a post about a security vulnerability in wiki on bugtraq a
couple of days ago, but it looked more like someone had figured out how
to edit pages (pointless considering a wiki is open anyway).

- --
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] what happened to asterisk wiki???

[Stephen Bosch]

Thanks for the pointers, Steve.

In general terms, it has always made me uncomfortable that the bulk of
Asterisk "documentation" is on a wiki operated by some rarely named
person only peripherally related to the project.

It would be nice to make this information available in a friendlier medium:

- a Mediawiki-based wiki engine;
- allowing mirroring
- Keep it focused just on Asterisk, rather than the entire field

Has anyone heard from the people who are operating the wiki? And who is
that? (I have asked this question before; my questions were met with a
snarky retort.)

Is *anything* actually happening with this, or can we kiss it and its
information goodbye? (Google cache has its limits.)

Everybody has an interest in getting this thing up and running again, be
they newbies or gurus; "Check the wiki" won't be an adequate response to
an FAQ anymore, and that's bound to drive somebody nuts.

-Stephen-

Steve Totaro wrote:
> It is down.  Two options, search in Google, click on cached and then
> click on cached text only at the top. 
> 
>  
> 
> This may also be helpful
> http://web.archive.org/web/*/http://www.voip-info.org
> 
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB
>  
> 
> 
> 
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Rizwan
> Hisham
> *Sent:* Wednesday, March 14, 2007 9:46 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] what happened to asterisk wiki???
> 
>  
> 
> Hi
> im trying access the www.voip-info.org 
> website since yesterday but i cant open it. google search diaplay
> correct search results but it doesnt open when i click the link. it
> displays a message about tcp error which says -->"There was a problem
> communicating with the server". I dont know what the problem is. I just
> want to ask whether their server is down or not and is everybody having
> the same problem?
> 
> -- 
> Regards
> Rizwan Hisham
> Software Engineer
> 
> !DSPAM:45f810e629411804284693!
> 
> 
> 
> 
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[asterisk-users] While the VoIP-Info.org site is down...

[Shane Breen]

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well: 
http://www.thetelecomdirectory.com You will be able to upload white papers, 
list your company in our directory, release press releases all for FREE.


Here is where you do all of the above: 
http://www.thetelecomdirectory.com/signup/signup.asp


If you want to see how The Telecom Directory ranks visit: 
http://www.alexa.com/search?q=thetelecomdirectory.com


Hopefully VoIP-Info will come back up but in the meantime use the site to 
its full potential. IT IS FREE.




- Original Message - 
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???



-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

JR Richardson wrote:

A friend of mine was on the site yesterday, late morning, when he
refreshed his screen, a banner came across the web page "VOIP SUCKS"
and then the site was no longer available.  I'm pretty sure the site
was compromised by some hacker trying to prove a point or make a
statement.  Not to throw stink on anyone or group, but maybe it was
someone from a competing open source VoIP project or one of the Big
Iron VoIP System Manufacturers.  Probably just some cracker with too
much time on their hands.  I feel like someone shot my dog, please get
the site back up as soon as possible.


There was a post about a security vulnerability in wiki on bugtraq a
couple of days ago, but it looked more like someone had figured out how
to edit pages (pointless considering a wiki is open anyway).

- --
Cheers,

Matt Riddell
Director
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iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
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[asterisk-users] While the VoIP-Info.org site is down...

[Shane Breen]

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well: 
http://www.thetelecomdirectory.com You will be able to upload white papers, 
list your company in our directory, release press releases all for FREE.


Here is where you do all of the above: 
http://www.thetelecomdirectory.com/signup/signup.asp


If you want to see how The Telecom Directory ranks visit: 
http://www.alexa.com/search?q=thetelecomdirectory.com


Hopefully VoIP-Info will come back up but in the meantime use the site to 
its full potential. IT IS FREE.




- Original Message - 
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???



-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

JR Richardson wrote:

A friend of mine was on the site yesterday, late morning, when he
refreshed his screen, a banner came across the web page "VOIP SUCKS"
and then the site was no longer available.  I'm pretty sure the site
was compromised by some hacker trying to prove a point or make a
statement.  Not to throw stink on anyone or group, but maybe it was
someone from a competing open source VoIP project or one of the Big
Iron VoIP System Manufacturers.  Probably just some cracker with too
much time on their hands.  I feel like someone shot my dog, please get
the site back up as soon as possible.


There was a post about a security vulnerability in wiki on bugtraq a
couple of days ago, but it looked more like someone had figured out how
to edit pages (pointless considering a wiki is open anyway).

- --
Cheers,

Matt Riddell
Director
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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
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Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

[Stephen Bosch]

Mojo with Horan & Company, LLC wrote:
> IIRC, you need an extension named 'callpark' in your extensions.conf
> that calls the ParkAndAnnounce application.
> 
> This should get you started:
> 
> exten =>
> callpark,1,ParkAndAnnounce(PARKED|600|Local/4${BRIDGEPEER:5:[EMAIL 
> PROTECTED]|incoming,s,1)
> 
> 
> in the CLI:
> Show Application ParkAndAnnounce
> for usage info.
> 
> The third field, in my case "Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]"
> is the channel to announce the parked call slot to.  In my case,
> extensions beginning with 1xx are the phones themselves, and extensions
> 4xx are the same phones but will make them auto-answer (like paging).
> You might have a better way to do this because this is a little cumbersome.
> 
> hope it helps!
> 
> Mojo

Hey, Mojo:

Thanks for this!

-Stephen-
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Re: [asterisk-users] TDM-400, Polycom SIP phones, and echo problems

[Matthew Fredrickson]

On Mar 12, 2007, at 7:12 PM, Stephen Bosch wrote:


Hi:

I am working on a new system with a TDM-400P card with three FXO 
modules

and one FXS module.

The system has been in place for a week. Users are complaining of echo
problems. I have noticed this echo myself. It varies in severity. It is
sometimes bad enough to make it difficult to converse, but the users
find it generally unacceptable. They miss their old phones, which just
worked. As you can imagine, this is not the way to get them excited
about this technology.

I have used the 1.4 branch version of fxotune to tune the card. These
are the echo statistics I get on the affected channels.


asterisk1 asterisk # fxotune -d -b 1 -w 1004
Dumping module /dev/zap/1
echo ratio = 0.0077 (85.5 / 11145.0)
Done!
asterisk1 asterisk # fxotune -d -b 3 -w 1004
Dumping module /dev/zap/3
echo ratio = 0.0296 (330.2 / 11145.0)
Done!


According to the wiki, both those echo ratios should be more than
sufficient to let the echo canceller handle the echo. Residual echo is
present, however. It doesn't interfere with conversation, but I could
see how it would be irritating, especially if you never had it before.
The call experience should get better, rather than worse -- especially
if we're using Polycom IP 650s.

Questions:

- What settings might I tweak to eliminate this remaining echo?
- Is there a hardware or software echo canceller that will do a more
thorough job of it?
- How is the HPEC in dealing with this echo?


It's as good (if not better) than using Octasic hardware echo 
cancellation.  I would definitely download it and install it.  And IIRC 
it comes free with the purchase of your TDM card from Digium.


Matthew Fredrickson

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RE: [asterisk-users] what happened to asterisk wiki???

[Steve Totaro]

I think Digium should host a wiki (keeping if vendor neutral of course).


This seems to be the most complete backup of voip-info.org but it is
fairly old http://web.archive.org/web/20051013074214/voip-info.org/wiki/
unless someone else spidered it more completely and recently. 

Even if VoIP-info comes back online, I think Digium or the Asterisk
documentation project should spider VoIP-info or at the least, the
snapshot from above and take over the effort.  

When was Paul Mahler's book published?  That was almost an exact cut and
paste of the wiki ;-)

It should not be run by someone who can just pull the plug one day.

Heck, with the introduction of version 1.4, start from scratch.  There
was a lot of junk mixed in with a lot of golden nuggets on Voip-info.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen Bosch
> Sent: Wednesday, March 14, 2007 4:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] what happened to asterisk wiki???
> 
> Thanks for the pointers, Steve.
> 
> In general terms, it has always made me uncomfortable that the bulk of
> Asterisk "documentation" is on a wiki operated by some rarely named
> person only peripherally related to the project.
> 
> It would be nice to make this information available in a friendlier
> medium:
> 
> - a Mediawiki-based wiki engine;
> - allowing mirroring
> - Keep it focused just on Asterisk, rather than the entire field
> 
> Has anyone heard from the people who are operating the wiki? And who
is
> that? (I have asked this question before; my questions were met with a
> snarky retort.)
> 
> Is *anything* actually happening with this, or can we kiss it and its
> information goodbye? (Google cache has its limits.)
> 
> Everybody has an interest in getting this thing up and running again,
be
> they newbies or gurus; "Check the wiki" won't be an adequate response
to
> an FAQ anymore, and that's bound to drive somebody nuts.
> 
> -Stephen-
> 
> Steve Totaro wrote:
> > It is down.  Two options, search in Google, click on cached and then
> > click on cached text only at the top.
> >
> >
> >
> > This may also be helpful
> > http://web.archive.org/web/*/http://www.voip-info.org
> >
> > Thanks,
> > Steve Totaro
> > http://www.asteriskhelpdesk.com
> > KB3OPB
> >
> >
> >

> >
> > *From:* [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] *On Behalf Of
*Rizwan
> > Hisham
> > *Sent:* Wednesday, March 14, 2007 9:46 AM
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Subject:* [asterisk-users] what happened to asterisk wiki???
> >
> >
> >
> > Hi
> > im trying access the www.voip-info.org 
> > website since yesterday but i cant open it. google search diaplay
> > correct search results but it doesnt open when i click the link. it
> > displays a message about tcp error which says -->"There was a
problem
> > communicating with the server". I dont know what the problem is. I
just
> > want to ask whether their server is down or not and is everybody
having
> > the same problem?
> >
> > --
> > Regards
> > Rizwan Hisham
> > Software Engineer
> >
> > !DSPAM:45f810e629411804284693!
> >
> >
> >

> >
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RE: [asterisk-users] While the VoIP-Info.org site is down...

[Steve Totaro]

Is it wise to use an outage to promote your business, not on the user's
list and not multiple times?  Put it in your signature or something ;-)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Shane Breen
> Sent: Wednesday, March 14, 2007 5:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] While the VoIP-Info.org site is down...
> 
> Feel free to use:  http://www.thetelecomdirectory.com/forum
> 
> If you register your company here as well:
> http://www.thetelecomdirectory.com You will be able to upload white
> papers,
> list your company in our directory, release press releases all for
FREE.
> 
> Here is where you do all of the above:
> http://www.thetelecomdirectory.com/signup/signup.asp
> 
> If you want to see how The Telecom Directory ranks visit:
> http://www.alexa.com/search?q=thetelecomdirectory.com
> 
> Hopefully VoIP-Info will come back up but in the meantime use the site
to
> its full potential. IT IS FREE.
> 
> 
> 
> - Original Message -
> From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, March 14, 2007 4:22 PM
> Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???
> 
> 
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> >
> > JR Richardson wrote:
> >> A friend of mine was on the site yesterday, late morning, when he
> >> refreshed his screen, a banner came across the web page "VOIP
SUCKS"
> >> and then the site was no longer available.  I'm pretty sure the
site
> >> was compromised by some hacker trying to prove a point or make a
> >> statement.  Not to throw stink on anyone or group, but maybe it was
> >> someone from a competing open source VoIP project or one of the Big
> >> Iron VoIP System Manufacturers.  Probably just some cracker with
too
> >> much time on their hands.  I feel like someone shot my dog, please
get
> >> the site back up as soon as possible.
> >
> > There was a post about a security vulnerability in wiki on bugtraq a
> > couple of days ago, but it looked more like someone had figured out
how
> > to edit pages (pointless considering a wiki is open anyway).
> >
> > - --
> > Cheers,
> >
> > Matt Riddell
> > Director
> > ___
> >
> > http://www.sineapps.com/news.php (Daily Asterisk News - html)
> > http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
> > -BEGIN PGP SIGNATURE-
> > Version: GnuPG v1.4.2 (MingW32)
> > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> >
> > iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
> > NV4lAmOkaISR79fBTjajGw8=
> > =u7sc
> > -END PGP SIGNATURE-
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Re: [asterisk-users] Zaptel version for asterisk 1.2.16

[Kevin P. Fleming]

John Novack wrote:
> In Digium's infinite wisdom, they have seen fir to have version numbers
> no longer match, and also NOT provide any sort of map to give the rest
> of us a clue as to what goes with what.
> 
> Probably to discourage wider use of the product?

I must say that is an excellent attitude and must win you a lot of
friends :-)

What we did was stop making 'artificial' releases of Zaptel and libpri
only to keep their version numbers in line with Asterisk version
numbers. There is no need nor value in having matching version numbers;
they are independent projects. If they were supposed to be released in
lock-step fashion, they'd all be a single project.

There is no need for any 'map'; any Asterisk 1.2.x release should be
usable with any Zaptel 1.2.x release, but of course we'd suggest using
the latest releases of both. There are no API changes or feature
additions (generally) in release branches, so frequently you can update
_only_ Asterisk if you are happy with the version of Zaptel you have
installed and running.
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Re: [asterisk-users] [G.729] Input Gain

[Kevin P. Fleming]

Victor Mateevitsi wrote:
> Can I configure the input gain of a context on sip.conf or maybe can I
> configure the input gain of g.729 encoded streams. In zapata.conf (or
> misdn.conf) I know there is the above option (rxgain and txgain).

No. Asterisk does not provide any facilities to adjust volume levels of
IP media streams.
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Re: [asterisk-users] what happened to asterisk wiki???

[Erik Anderson]

On 3/14/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:


It would be nice to make this information available in a friendlier medium:

- a Mediawiki-based wiki engine;
- allowing mirroring
- Keep it focused just on Asterisk, rather than the entire field


Well put.  I made the (unfortunate) decision to use tikiwiki (what's
currently running voip-info.org) at work for an internal wiki a few
years ago.  Man, what a crappy piece of software.  Luckily, we've
since move to greener pastures with Mediawiki.

Anyway - I don't mind so much having a general-purpose VoIP wiki, but
I *do* resonate with your desires to have it be easily-mirrorable.  I
believe you can download nightly snapshots of the wikipedia
databases...if the voip-info.org folks offerred a similar service, I
know there would be many people (myself include) who would love to
mirror the data, making it publicly available, thereby reducing some
of the load on the main server.

For some reason, I'm doubtful this will ever happen, but hey, we can
always hope, right?

Here's to v-i.org coming back soon.

-Erik Anderson

--
Erik Anderson
http://andersonfam.org
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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Shane Breen]

Sorry about that. I figured since it is a FREE site it was no biggie.

Thanks,

Shane Breen *The Telecom Directory * Office: 404-797-6633
[EMAIL PROTECTED] * www.TheTelecomDirectory.com

The Telecom Directory is the only complete A to Z interactive website for 
all telecommunication related information, services, solutions and products.


- Original Message - 
From: "Steve Totaro" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, March 14, 2007 5:57 PM
Subject: RE: [asterisk-users] While the VoIP-Info.org site is down...


Is it wise to use an outage to promote your business, not on the user's
list and not multiple times?  Put it in your signature or something ;-)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Shane Breen
Sent: Wednesday, March 14, 2007 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] While the VoIP-Info.org site is down...

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well:
http://www.thetelecomdirectory.com You will be able to upload white
papers,
list your company in our directory, release press releases all for

FREE.


Here is where you do all of the above:
http://www.thetelecomdirectory.com/signup/signup.asp

If you want to see how The Telecom Directory ranks visit:
http://www.alexa.com/search?q=thetelecomdirectory.com

Hopefully VoIP-Info will come back up but in the meantime use the site

to

its full potential. IT IS FREE.



- Original Message -
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???


> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> JR Richardson wrote:
>> A friend of mine was on the site yesterday, late morning, when he
>> refreshed his screen, a banner came across the web page "VOIP

SUCKS"

>> and then the site was no longer available.  I'm pretty sure the

site

>> was compromised by some hacker trying to prove a point or make a
>> statement.  Not to throw stink on anyone or group, but maybe it was
>> someone from a competing open source VoIP project or one of the Big
>> Iron VoIP System Manufacturers.  Probably just some cracker with

too

>> much time on their hands.  I feel like someone shot my dog, please

get

>> the site back up as soon as possible.
>
> There was a post about a security vulnerability in wiki on bugtraq a
> couple of days ago, but it looked more like someone had figured out

how

> to edit pages (pointless considering a wiki is open anyway).
>
> - --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.2 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
> NV4lAmOkaISR79fBTjajGw8=
> =u7sc
> -END PGP SIGNATURE-
> ___
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>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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RE: [asterisk-users] Packetization Rate

[Dan Austin]

Matt wrote:
> To my knowledge, Asterisk's packetization rate is hard 
> coded at 30ms.  If I wanted to, where in the code could
> I go to change it to 20ms.   Is there anything bad that 
> might happen if I change it (asterisk related)?

You don't mention what version you are using, but 1.4 does
support alternate framing (packetization) options on a per
codec basis.

The feature originally was based on SVN trunk when it was
still close to 1.2, but I would not want to try to backport
and support it.


Dan
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Re: [asterisk-users] Polycom call parking feature and Asterisk call parking

[Noah Miller]

Hi Stephen -


I want to make parking calls easier for my hard-working users. Is there
a way to make the Polycom call park feature work with Asterisk?

In case it just works out of the box, I haven't tried it yet; but the
"call park" feature isn't enabled on the Polycom phones by default.


If you want to make it even easier, you can make parking work with a
single DTMF press.  I believe 1.4 has this feature by default using
the "kK" flags for dial(), and setting it up in features.conf.

If you're still running 1.2.x, I can provide you with a patch.  It's a
bit hack-ish, but I've been using it production for over a year now.

- Noah
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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Davis Sylvester III]

Shane Breen wrote:

Sorry about that. I figured since it is a FREE site it was no biggie.

Thanks,

Shane Breen *The Telecom Directory * Office: 404-797-6633
[EMAIL PROTECTED] * www.TheTelecomDirectory.com

The Telecom Directory is the only complete A to Z interactive website 
for all telecommunication related information, services, solutions and 
products.


- Original Message - From: "Steve Totaro" 
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, March 14, 2007 5:57 PM
Subject: RE: [asterisk-users] While the VoIP-Info.org site is down...


Is it wise to use an outage to promote your business, not on the user's
list and not multiple times?  Put it in your signature or something ;-)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Shane Breen
Sent: Wednesday, March 14, 2007 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] While the VoIP-Info.org site is down...

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well:
http://www.thetelecomdirectory.com You will be able to upload white
papers,
list your company in our directory, release press releases all for

FREE.


Here is where you do all of the above:
http://www.thetelecomdirectory.com/signup/signup.asp

If you want to see how The Telecom Directory ranks visit:
http://www.alexa.com/search?q=thetelecomdirectory.com

Hopefully VoIP-Info will come back up but in the meantime use the site

to

its full potential. IT IS FREE.



- Original Message -
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???


> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> JR Richardson wrote:
>> A friend of mine was on the site yesterday, late morning, when he
>> refreshed his screen, a banner came across the web page "VOIP

SUCKS"

>> and then the site was no longer available.  I'm pretty sure the

site

>> was compromised by some hacker trying to prove a point or make a
>> statement.  Not to throw stink on anyone or group, but maybe it was
>> someone from a competing open source VoIP project or one of the Big
>> Iron VoIP System Manufacturers.  Probably just some cracker with

too

>> much time on their hands.  I feel like someone shot my dog, please

get

>> the site back up as soon as possible.
>
> There was a post about a security vulnerability in wiki on bugtraq a
> couple of days ago, but it looked more like someone had figured out

how

> to edit pages (pointless considering a wiki is open anyway).
>
> - --
> Cheers,
>
> Matt Riddell
> Director
> ___
>
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://wap.sineapps.com (Daily Asterisk News for your cellphone)
> http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.2 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
> NV4lAmOkaISR79fBTjajGw8=
> =u7sc
> -END PGP SIGNATURE-
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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If needed I can put up a wiki for asterisk support, and allow 
mirroring.  I know we have to start over from ground zero, but it gets 
us back online with our life line.  Let me know if you guys want me to 
proceed.  There would be no charge for us hosting the site.


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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Stephen Bosch]

Shane Breen wrote:
> Sorry about that. I figured since it is a FREE site it was no biggie.
> 
> Thanks,
> 
> Shane Breen

Well, Shane, nothing is ever really FREE, and when someone offers
something FREE, there is almost always a catch, and on the Internet that
catch tends to come in the form of SPAM.

Spam that looks not unlike your e-mails to the list.

What we want is for the wiki to be back up because it contains useful
information. We can get advertising anywhere.

-Stephen-

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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Stephen Bosch]

Davis Sylvester III wrote:
> If needed I can put up a wiki for asterisk support, and allow
> mirroring.  I know we have to start over from ground zero, but it gets
> us back online with our life line.  Let me know if you guys want me to
> proceed.  There would be no charge for us hosting the site.

This is one of those cases where deeds speak louder than words. If you
have a server available, put it up. Tell us where it is and we can start
putting stuff from the Google cache in it.

If you can, make it use Mediawiki. The editing interface is much friendlier.

It is looking increasingly like voip-info.org is history, anyway -- and
you're right, we do need something.

-Stephen-

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Re: [asterisk-users] what happened to asterisk wiki???

[Stephen Bosch]

Steve Totaro wrote:
> I think Digium should host a wiki (keeping if vendor neutral of course).

It escapes me why they haven't done this sooner. It would only help them
sell product.

> This seems to be the most complete backup of voip-info.org but it is
> fairly old http://web.archive.org/web/20051013074214/voip-info.org/wiki/
> unless someone else spidered it more completely and recently. 

What about the Google caches that you pointed out? Getting the
information would be tedious, but not impossible; we just have to act
quickly.

Our business doesn't have any spare servers available, and we're short
colo space anyway; I don't know where we'd put the thing if we did.

I would like to help, though -- we operate a Mediawiki for our internal
documentation, and it's brilliant. Very flexible, makes publishing a
document set quickly very easy.

> Even if VoIP-info comes back online, I think Digium or the Asterisk
> documentation project should spider VoIP-info or at the least, the
> snapshot from above and take over the effort.  

The deafening silence on this matter suggests that it is not coming back.

> When was Paul Mahler's book published?  That was almost an exact cut and
> paste of the wiki ;-)
> 
> It should not be run by someone who can just pull the plug one day.

Hear hear. It was only a matter of time before this happened.

> Heck, with the introduction of version 1.4, start from scratch.  There
> was a lot of junk mixed in with a lot of golden nuggets on Voip-info.

Well, we still need 1.2.x docs, but I'm all for starting from scratch.
There was some seriously ancient info in there, and tiki is so
atrociously arcane it discourages people from making improvements.

-Stephen-

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Re: [asterisk-users] what happened to asterisk wiki???

[Andy Brezinsky]

I work for a conferencing and telecom company and have used
voip-info.org more times than I can count for both VoIP and asterisk
reasons.  We've used asterisk internally and have greatly benefited from
it.

We're willing to host the Wiki with database dumps available for people
to download on a regular basis (to ensure continuity should something
happen on our end) and dedicate necessary resources to hosting the
application in our Milwaukee or Chicago locations.

I've setup a MediaWiki instance at a temporary url (wiki.mixmeeting.com)
until a domain name is decided upon should the community in general wish
to use our hosting.

-- 
Andy Brezinsky
Chief Engineer
Brevient Technologies

Office: (414) 944-0162 x1029
Direct: (414) 944-0190


On Wed, 2007-03-14 at 17:03 -0600, Stephen Bosch wrote:
> Steve Totaro wrote:
> > I think Digium should host a wiki (keeping if vendor neutral of course).
> 
> It escapes me why they haven't done this sooner. It would only help them
> sell product.
> 
> > This seems to be the most complete backup of voip-info.org but it is
> > fairly old http://web.archive.org/web/20051013074214/voip-info.org/wiki/
> > unless someone else spidered it more completely and recently. 
> 
> What about the Google caches that you pointed out? Getting the
> information would be tedious, but not impossible; we just have to act
> quickly.
> 
> Our business doesn't have any spare servers available, and we're short
> colo space anyway; I don't know where we'd put the thing if we did.
> 
> I would like to help, though -- we operate a Mediawiki for our internal
> documentation, and it's brilliant. Very flexible, makes publishing a
> document set quickly very easy.
> 
> > Even if VoIP-info comes back online, I think Digium or the Asterisk
> > documentation project should spider VoIP-info or at the least, the
> > snapshot from above and take over the effort.  
> 
> The deafening silence on this matter suggests that it is not coming back.
> 
> > When was Paul Mahler's book published?  That was almost an exact cut and
> > paste of the wiki ;-)
> > 
> > It should not be run by someone who can just pull the plug one day.
> 
> Hear hear. It was only a matter of time before this happened.
> 
> > Heck, with the introduction of version 1.4, start from scratch.  There
> > was a lot of junk mixed in with a lot of golden nuggets on Voip-info.
> 
> Well, we still need 1.2.x docs, but I'm all for starting from scratch.
> There was some seriously ancient info in there, and tiki is so
> atrociously arcane it discourages people from making improvements.
> 
> -Stephen-
> 
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Re: [asterisk-users] Patton 1400

[Jean-Louis curty]

Hi,
I managed to install my patton gateway but i did not succeed to pass the
caller id to the sip phone on incoming calls ...
instead i see call from 105 ( which is the sip client extension of the
patton )

do you know the way to pass the caller id of the caller to the ip phone iso
the gw extension on incoming calls ?
thanks !
jean-louis


2006/11/4, Guido Hecken <[EMAIL PROTECTED]>:


Hi Kevin,

you have to create a gateway in the Smart Node:

gateway sip sip
bind interface eth1 router

service default
   domain gwsnettech.local
   realm gwsnettech.local
   authentication isdngw2 password huffvtzddzdjkhuztztufuz== encrypted
default
   default-server hallinux2.gwsnettech.local 5060 loose-router
   registrar hallinux2.gwsnettech.local 5060

In sip.conf, something like this:

[isdngw2]
type=friend
username=isdngw2
secret=the_unencrypted_password_from _above
host=192.168.161.135
;host=dynamic
fromuser=gwsnettech
fromdomain=gwsnettech.local
nat=no
context=isdngw2-in
canreinvite=no

If you need a complete running config, I can send it to you offlist.

Hope, it helps...


Guido


Von: Kevin Withnall [mailto:[EMAIL PROTECTED]
Gesendet: Samstag, 4. November 2006 05:58
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Patton 1400

I have a patton 1400 setup to handle the bri interface. As a trixbox user,
I
wanted a sip trunk rather than having to re-compile bri support into
trixbix.

Anyway, I have it working now so that asterisk can make calls and they are
passed properly to the telephone network. Incoming calls however are
another
matter. I have (after turning on cli debug in the 1400) determined that
its
getting stuck in the routing system. I don't know what destination to make
to get it to sip connect to the asterisk box.

Ive tried making an interface that has a remote address of the asterisk
box
but that doesn't work. Can someone send me a config for anything they have
done that is similar to this ? I believe the Patton 1200 is also the same
unit apart from extra ports.

Any help would be greatly appreciated.

Regards
Kevin

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Re: [asterisk-users] what happened to asterisk wiki???

[James Coberly]

We also would like to offer hosting and/or mirroring until VOIP-INFO
hopefully gets back on it's feet.
We have MediaWiki setup at www.xmc.com/wiki and will also temp in dns
voipwiki.xmc.com direct.

It is there if the community wants to use it also.


On Wed, 2007-03-14 at 18:26 -0500, Andy Brezinsky wrote:

> I work for a conferencing and telecom company and have used
> voip-info.org more times than I can count for both VoIP and asterisk
> reasons.  We've used asterisk internally and have greatly benefited from
> it.
> 
> We're willing to host the Wiki with database dumps available for people
> to download on a regular basis (to ensure continuity should something
> happen on our end) and dedicate necessary resources to hosting the
> application in our Milwaukee or Chicago locations.
> 
> I've setup a MediaWiki instance at a temporary url (wiki.mixmeeting.com)
> until a domain name is decided upon should the community in general wish
> to use our hosting.
> 
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[asterisk-users] Current voip-info.org Status

[David Schardin]

News for everyone on this.

Recently found out that this site is down due to a hardware failure.  
Hard Drives in the RAID array failed and currently the problem is  
being addressed as fast as possible. Hopes are that voip-info.org  
will be operational again sometime tomorrow afternoon.


Hope this helps alleviate some of the concerns in the VoIP community  
at this time as well as alleviate some of the questions being sent to  
the people at voip-info.org.


Sincerely,
Jessee J. Holmes
Atacomm Corporation
http://voipstore.atacomm.com
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[asterisk-users] voip-info.org status update

[James H Thompson]

A short status update:

Yesterday 3 of the 4 disk drives in the RAID array on the server that hosts 
voip-info.org failed.

The coloprovider is currently working to replace the drives and I'm hoping that 
the site returns to service soon.
Tomorrow is looking most likely.

I'd like to thank all those that have called or emailed to offer help and/or 
encouragement.

I will definately be looking for an easy way to create a mirror site once 
voip-info.org is back up.
This is made difficult by the dynamic nature of the site, but its been on my 
list of things to do for a while now.

Thanks for using voip-info.org!

[EMAIL PROTECTED] 





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[asterisk-users] Cisco 7912

[Matt Putnam]

I have 3 cisco 7912 that all stoped working at the same time on sunday.
There is nothing on the display and the menu and hold buttons are lit.
Resteing produces the same results the phone dosent respond. Anyone have an
idea how to fix this or if it can even be fixed. Ive done some searching
online and havent found anything useful any sugestions?
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RE: [asterisk-users] While the VoIP-Info.org site is down...

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen Bosch
> Sent: Wednesday, March 14, 2007 5:51 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] While the VoIP-Info.org site is down...
> 
> Davis Sylvester III wrote:
> > If needed I can put up a wiki for asterisk support, and allow
> > mirroring.  I know we have to start over from ground zero, but it
gets
> > us back online with our life line.  Let me know if you guys want me
to
> > proceed.  There would be no charge for us hosting the site.
> 
> This is one of those cases where deeds speak louder than words. If you
> have a server available, put it up. Tell us where it is and we can
start
> putting stuff from the Google cache in it.
> 
> If you can, make it use Mediawiki. The editing interface is much
> friendlier.
> 
> It is looking increasingly like voip-info.org is history, anyway --
and
> you're right, we do need something.
> 
> -Stephen-
> 

Anyone that thinks is will not come back online has not been around the
asterisk community very long.  This was the regular (2-3 times a week).
It would go down for a day, a couple days, a couple hours, and then it
would come back up.

If it really was hacked, then lets hope someone was doing a backup.  If
not, spidering the internet archive will get a good deal of it back.

Thanks,
Steve

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Re: [asterisk-users] TDM-400, Polycom SIP phones, and echo problems

[Ira]

At 02:47 PM 3/14/2007, you wrote:

- How is the HPEC in dealing with this echo?


Well, I don't have much of a system, but since I installed the HPEC 
there has been no echo on my POTs lines.


Ira 


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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Al Bochter]

So does anyone know when Voip-info.org will be back up?

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email



Steve Totaro wrote:


Is it wise to use an outage to promote your business, not on the user's
list and not multiple times?  Put it in your signature or something ;-)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB


 


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Shane Breen
Sent: Wednesday, March 14, 2007 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] While the VoIP-Info.org site is down...

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well:
http://www.thetelecomdirectory.com You will be able to upload white
papers,
list your company in our directory, release press releases all for
   


FREE.
 


Here is where you do all of the above:
http://www.thetelecomdirectory.com/signup/signup.asp

If you want to see how The Telecom Directory ranks visit:
http://www.alexa.com/search?q=thetelecomdirectory.com

Hopefully VoIP-Info will come back up but in the meantime use the site
   


to
 


its full potential. IT IS FREE.



- Original Message -
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???


   


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

JR Richardson wrote:
 


A friend of mine was on the site yesterday, late morning, when he
refreshed his screen, a banner came across the web page "VOIP
   


SUCKS"
 


and then the site was no longer available.  I'm pretty sure the
   


site
 


was compromised by some hacker trying to prove a point or make a
statement.  Not to throw stink on anyone or group, but maybe it was
someone from a competing open source VoIP project or one of the Big
Iron VoIP System Manufacturers.  Probably just some cracker with
   


too
 


much time on their hands.  I feel like someone shot my dog, please
   


get
 


the site back up as soon as possible.
   


There was a post about a security vulnerability in wiki on bugtraq a
couple of days ago, but it looked more like someone had figured out
 


how
 


to edit pages (pointless considering a wiki is open anyway).

- --
Cheers,

Matt Riddell
Director
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
NV4lAmOkaISR79fBTjajGw8=
=u7sc
-END PGP SIGNATURE-
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Inbound (clean). Database: 000723-2, 03/14/2007 - 3/14/2007 6:33:59 PM




 

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[asterisk-users] Inbound PSTN CLID irratic with A200

[Wireless]

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and 
yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with 
this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma 
support suggested changing the RXGAIN in zapata.conf and it worked. Now using 
Trixbox 2.0 and have upgraged to latest software

Asterisk 1.2.16
Zaptel 1.2.15
Wanpipe-2.3.4-7

and CLID is back to being a bit random again, with or without HPEC. I've 
fiddled with rxgain but no improvement. I've got ADSL on this line too and have 
read that this can be an issue but I know that it has worked well in the 
passed, and the ADSL and hardware has not changed

Odd thing is that if I reboot the Asterisk box the the 1st call that comes in 
always has the correct CLID

Anyone have any ideas?

Thanks

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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Germán Aracil Boned]

I don't know, but, I can put a server for mirror this page.
This page is a very good tool. I can put a mirror for this on Europe.

regards

Al Bochter escribió:

So does anyone know when Voip-info.org will be back up?

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email



Steve Totaro wrote:

Is it wise to use an outage to promote your business, not on the user's
list and not multiple times?  Put it in your signature or something ;-)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Shane Breen
Sent: Wednesday, March 14, 2007 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] While the VoIP-Info.org site is down...

Feel free to use:  http://www.thetelecomdirectory.com/forum

If you register your company here as well:
http://www.thetelecomdirectory.com You will be able to upload white
papers,
list your company in our directory, release press releases all for


FREE.
  

Here is where you do all of the above:
http://www.thetelecomdirectory.com/signup/signup.asp

If you want to see how The Telecom Directory ranks visit:
http://www.alexa.com/search?q=thetelecomdirectory.com

Hopefully VoIP-Info will come back up but in the meantime use the site


to
  

its full potential. IT IS FREE.



- Original Message -
From: "Matt Riddell (NZ)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, March 14, 2007 4:22 PM
Subject: Re: [asterisk-users] RE: what happened to asterisk wiki???




-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

JR Richardson wrote:
  

A friend of mine was on the site yesterday, late morning, when he
refreshed his screen, a banner came across the web page "VOIP


SUCKS"
  

and then the site was no longer available.  I'm pretty sure the


site
  

was compromised by some hacker trying to prove a point or make a
statement.  Not to throw stink on anyone or group, but maybe it was
someone from a competing open source VoIP project or one of the Big
Iron VoIP System Manufacturers.  Probably just some cracker with


too
  

much time on their hands.  I feel like someone shot my dog, please


get
  

the site back up as soon as possible.


There was a post about a security vulnerability in wiki on bugtraq a
couple of days ago, but it looked more like someone had figured out
  

how
  

to edit pages (pointless considering a wiki is open anyway).

- --
Cheers,

Matt Riddell
Director
___

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http://wap.sineapps.com (Daily Asterisk News for your cellphone)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67
NV4lAmOkaISR79fBTjajGw8=
=u7sc
-END PGP SIGNATURE-
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Inbound (clean). Database: 000723-2, 03/14/2007 - 3/14/2007 6:33:59 PM




  




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Re: [asterisk-users] Re: Asterisknow with video and X-Lite not quite working

[dave cantera]

benedikt,
try putting these (or your version of these) in the sip.conf [general]
heading.   it was suggested to me before, that what is general should
go in the general section, what is specific to a particular extension
should go in the specific extension section.  also, I put a ton of
options in my sip.conf, things wouldn't work as I expected.  I guess I
didn't understand all the options correctly.  took it all out, entered
just a few options, whalla, it worked.  KISS
[general]
  context=default
  bindport=5060
  bindaddr=0.0.0.0
  srvlookup=yes
  dtmfmode=rfc2833
  videosupport=yes
  maxcallbitrate=384
  nat=yes
  canreinvite=no
  allowsubscribe=yes
  notifyringing = yes
  limitonpeers=yes
  disallow=all
  allow=ulaw
  allow=gsm

[401]
  type=friend
  callerid=x401 <(800)000-0401>
  secret=1234
  qualify=5000
  nat=no
  host=10.10.15.41
  context=inbound-video
  allow=h264
  mailbox=401
  
voicemail.conf  - don't forget this one!
  
401 => 1234,x401
User,[EMAIL PROTECTED],[EMAIL PROTECTED]

remember, I reloaded 1.4.0 over *now with libpri, zaptel,
asterisk-addons too.

my x-lite version is:

© 2004 Xten Networks, Inc. All rights reserved.
X-Lite release 1103m build stamp 14262

it has a diagnostic log, you might want to look at that to see what
codecs the endpoints are negociating to send...  would be similar to
below.  also the * debugging as below.
From: "asterisk"
;tag=as6a0be506
  To: 
  Contact: 
  Call-ID: [EMAIL PROTECTED]
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX
  Max-Forwards: 70
  Date: Thu, 15 Mar 2007 01:10:57 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
  Supported: replaces
  Content-Length: 0

-and-
--- (15 headers 11 lines) ---
  Sending to 192.168.15.103 : 5060 (no NAT)
  Using INVITE request as basis request -
[EMAIL PROTECTED]
  Found user '300'
  Found RTP audio format 0
  Found RTP audio format 8
  Found RTP audio format 18
  Found RTP audio format 101
  Peer audio RTP is at port 192.168.15.103:2226
  Found description format PCMU for ID 0
  Found description format PCMA for ID 8
  Found description format G729 for ID 18
  Found description format telephone-event for ID 101
  Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
  Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x1 (telephone-event), combined - 0x1 (telephone-event)

the above is * debugging peer 301
I would imagine that you would find your h.263(p) codec in your debug
output... this is from a polycom 301, no video codec...
the users.conf file doesn't, IMHO, to be working reliably, your mileage
may vary...  that is why I reloaded over with 1.4.0...
hope that helps...
daveC


Benedikt Franz wrote:

  Hi Dave,

yes, Audio is fine, but no video. And as far as I can see, X-Lite (running 3.0 build 34025 here, all Clients have exactly the same version) supports only the h.263 and h.263p video codecs. But I am not quite sure if I enabled these codecs properly. For *now, I have put the allow-lines into the users.conf, for instance, heres my setup (I cencored out email and secret):

[6510]
fullname = Benedikt Franz
secret = ...
email = ...
cid_number = 6510
zapchan = 
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = yes
hasmanager = yes
callwaiting = yes
threewaycalling = yes
mailbox = 6510
hasagent = no
group = 
host = dynamic
registersip = yes
registeriax = yes
allow = h263
allow = h263p
canreinvite = yes


I am not sure if that is a NAT problem, since all users are either on the local area network, or connected through VPN (I have not tested video with those yet, though), however, I will try that.
  


-- 
Building Strong Relationships w/ Intelligent Customer Service
--

Interlocking Business Solutions, LLC
856-380-0894 x5000




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Re: [asterisk-users] Zaptel version for asterisk 1.2.16

[John Novack]

Kevin P. Fleming wrote:

John Novack wrote:
  

In Digium's infinite wisdom, they have seen fir to have version numbers
no longer match, and also NOT provide any sort of map to give the rest
of us a clue as to what goes with what.

Probably to discourage wider use of the product?



I must say that is an excellent attitude and must win you a lot of
friends :-)
  

Thank you.
Simply expressing the feelings of many who either choose not to express 
it openly or when they do you can't hear them.

What we did was stop making 'artificial' releases of Zaptel and libpri only to 
keep their version numbers in line with Asterisk version numbers. There is no 
need nor value in having matching version numbers;
  
You are certainly entitled to your opinion, but it results in a GREAT 
deal of confusion for some attempting to plan an install.
Not EVERYONE knows the product as well as you, and a simple statement as 
you made in the following paragraphs would go a very long way to solve 
this very small problem



they are independent projects. If they were supposed to be released in 
lock-step fashion, they'd all be a single project.

There is no need for any 'map'; any Asterisk 1.2.x release should be usable 
with any Zaptel 1.2.x release, but of course we'd suggest using the latest 
releases of both. There are no API changes or feature additions (generally) in 
release branches, so frequently you can update _only_ Asterisk if you are happy 
with the version of Zaptel you have installed and running.

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Re: [asterisk-users] Packetization Rate

[Matt]

I am currently using 1.2 and can not upgrade to 1.4 until it becomes stable
and we have done much testing with it.

Obviously somewhere in the asterisk code 30ms must be coded... is it set in
just one place, and if so can I set that to 20ms?

On 3/14/07, Dan Austin <[EMAIL PROTECTED]> wrote:


Matt wrote:
> To my knowledge, Asterisk's packetization rate is hard
> coded at 30ms. If I wanted to, where in the code could
> I go to change it to 20ms. Is there anything bad that
> might happen if I change it (asterisk related)?

You don't mention what version you are using, but 1.4 does
support alternate framing (packetization) options on a per
codec basis.

The feature originally was based on SVN trunk when it was
still close to 1.2, but I would not want to try to backport
and support it.


Dan
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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Darryl Ross]

[Al Bochter wrote on 15/03/2007 12:25 PM]:
> So does anyone know when Voip-info.org will be back up?

There is a message on the list from James Thompson with the subject
"voip-info.org status update" saying it suffered a major hard drive
crash and should be back tomorrow.

Looking at the headers that message was sent two hours before yours.

Regards
Darryl
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Re: [asterisk-users] Cisco 7912

[Hermann Wecke]

Matt Putnam wrote:

anything useful any sugestions?


Are they requesting anything via TFTP? Do you have the full tftp files 
ready?

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Re: [asterisk-users] While the VoIP-Info.org site is down...

[Matt]

Yikes.. you'd think a server would be running RAID.

At any rate.. Please feel free to visit http://www.voip-wiki.us

I have set this up to be able to hold information for the Asterisk
community.  I will also gladly allow others to mirror it.

It is sitting in a climate controlled data center in Central PA on a server
with RAID.  Additionally, it is at the end of 95Megabytes/second on a BGP
redundant connection.

Please feel free to use it, if the community feels it can be useful...
additionally, I would love to setup some rsync mirrors with others so that
we can have redundant backups of this very valuable information.

On 3/14/07, Darryl Ross <[EMAIL PROTECTED]> wrote:


[Al Bochter wrote on 15/03/2007 12:25 PM]:
> So does anyone know when Voip-info.org will be back up?

There is a message on the list from James Thompson with the subject
"voip-info.org status update" saying it suffered a major hard drive
crash and should be back tomorrow.

Looking at the headers that message was sent two hours before yours.

Regards
Darryl
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Re: [asterisk-users] Linksys not Ringing

[dave cantera]

jason,
shouldn't there be an answer in there somewhere?... like...
[inbound-sip]
  exten => 300,1,Wait(1)
  exten => 300,n,Answer()
  exten => 300,n,NoOp(${EXTEN})
  exten => 300,n,NoOp(${CALLERID})
  exten => 300,n,Dial(SIP/300,15)
  exten => 300,n,VoiceMailMain
  exten => 300,n,Hangup()

daveC

Jason Walker wrote:
I
have 2 linksys SIP phones SPA-942
  
I have  a dialplan of
  
  
exten => 144,1,Wait(1)
  
exten => 144,2,Dial(Sip/phil,20)
  
exten => 144,3,Voicemail([EMAIL PROTECTED],u)
  
  
The CLI looks like this when I dial 144
  
  
-- Executing Wait("IAX2/JASONSERVER-9", "1") in new stack
  
   -- Executing Dial("IAX2/JASONSERVER-9", "Sip/phil|20") in new stack
  
   -- Called phil
  
   -- Nobody picked up in 2 ms
  
   -- Executing VoiceMail("IAX2/JASONSERVER-9", "[EMAIL PROTECTED]|u") in
new stack
  
   -- Playing 'vm-theperson' (language 'en')
  
  
It is registered and will make calls but I never get the
  
  -- SIP/phil is ringing
  
  
This happening on my 2 linksys phones only
  
  
Jason
  
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Re: [asterisk-users] Cisco 7912

[Matt Putnam]

I didnt have them on tftp files they were all manualy configured. They are
not trying to request anything they have the tftp server address but are not
requesting any files. It should start up and look for a vlan but its not
even doing that it does nothing when i plug it in just a blank screen and
the red and green leds on the hold and menu buttons are lit.

On 3/14/07, Hermann Wecke <[EMAIL PROTECTED]> wrote:


Matt Putnam wrote:
> anything useful any sugestions?

Are they requesting anything via TFTP? Do you have the full tftp files
ready?
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Re: [asterisk-users] Packetization Rate

[Luki]

Obviously somewhere in the asterisk code 30ms must be coded... is it set in
just one place, and if so can I set that to 20ms?


The default is 20 ms for most (all?) codecs. It's in rtp.c, where
ast_rtp_write() creates a new smoother.

--Luki
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[asterisk-users] Voip-Wiki Site Information

[Matt]

Community,
I have put up www.voip-wiki.us
My apologies to our fellow Asteristians outside the us... this was the only
easy domain available.

At any rate, feel free to populate the database / wiki, and I will be more
then happy to have and help others mirror this site so we can have
duplicates, Heaven forbid, something should happen to the server or Internet
connection.
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Re: [asterisk-users] Cisco 7912

[Tom Lynn]

Do they appear to have failed as a result of Daylight Savings time?

On 3/14/07, Matt Putnam <[EMAIL PROTECTED]> wrote:


I didnt have them on tftp files they were all manualy configured. They are
not trying to request anything they have the tftp server address but are not
requesting any files. It should start up and look for a vlan but its not
even doing that it does nothing when i plug it in just a blank screen and
the red and green leds on the hold and menu buttons are lit.

On 3/14/07, Hermann Wecke <[EMAIL PROTECTED]> wrote:
>
> Matt Putnam wrote:
> > anything useful any sugestions?
>
> Are they requesting anything via TFTP? Do you have the full tftp files
> ready?
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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