[asterisk-users] Redundancy

[Khaled]

Dears

Do any one have an idea to make a redundant plan for asterisk ,if a call
established  between two points and the server  interface became down ,do we
you have an idea how to let the call established till the collie or the
caller  hang-up.

Regards

   




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[asterisk-users] caller id not transferred to SIP device

[Tobias Unsleber]

Hello,

I'm wondering why asterisk is not transferring the callerid to the sip device. 
Scenario as follows:

sangoma --- zaptel --- asterisk --- sip --- SIP-Device

zaptel is reporting the callerid, but in the sip packages the sip-address 
shows unknown as user part, as this sip debug package shows:

Executing Dial(Zap/62-1, SIP/123|25|d) in new stack
We're at 172.31.253.80 port 10460
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.11.47:2075:
INVITE sip:[EMAIL PROTECTED]:2075;line=gv8x1x75 SIP/2.0
Via: SIP/2.0/UDP 172.31.253.80:5060;branch=z9hG4bK5e96f554;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as14f7c144
To: sip:[EMAIL PROTECTED]:2075;line=gv8x1x75
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Jan 2007 08:58:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 265

--

Versions:

asterisk 1.2.14
zaptel 1.2.12
linux 2.6.15.7
asterisk addons 1.2.4


SIP-Device:

I set CallingPress to allowed also, no effect. I think this is for the 
outgoing caller id presentation. (?)

SIP device config(sip show peer)
 * Name   : 123
  Secret   : Set
  MD5Secret: Not set
  Context  : wahlplan_international
  Subscr.Cont. : Not set
  Language : de
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 5
  Pickupgroup  : 5
  Mailbox  : 123
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Dynamic  : Yes
  Callerid :  
  Expire   : 3010
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 192.168.11.47 Port 2069
  Defaddr-IP  : 0.0.0.0 Port 2069
  Def. Username: 123
  SIP Options  : (none)
  Codecs   : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Status   : Unmonitored
  Useragent: snom360/6.5.1
  Reg. Contact : sip:[EMAIL PROTECTED]:2069;line=h9dxgpnb
-- 
Tobias Unsleber
VoIP Consultant

focus::voip GmbH
http://www.focus-voip.de

Hausadresse:
Robert-Koch-Strasse 9
D-64331 weiterstadt

Postfach 10 01 21
D-64201 Darmstadt

Tel.: +49 61 51 / 90 67 - 256
FAX : +49 61 51 / 90 67 - 299
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Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes

[Thomas Kenyon]

Brad Templeton wrote:

On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote:

Brad Templeton wrote:


For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.

Unless bandwidth between the * servers is a concern, then you're better 
off keeping the link between servers as IAX. (preferably trunked)


The bandwidth of the audio stream dwarfs the bandwidth of signalling
traffic by orders of mangitude.   So in fact, I think this is exactly
wrong.  If bandwidth to or between the servers is a concern, that's
where you most want to not be in the audio path.


But if you have multiple RTP streams emnbedded in an IAX trunk, then the 
IP overhead is significantly reduced.


AFAIK video should work for IAX2, there is explicit support for it. 
(unlike h323).


Asterisk will only be able to pass the raw RTP traffic though, since it 
doesn't have any video codecs (just format definitions).



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Re: [asterisk-users] ooh323c calls

[Michel]

Hello,

Thanks you for your reply.

The number in context test of asterisk B is  150.

exten = 15,n,Dial(OOH323/150/mypeer1);or  exten = 
15,n,Dial(OOH323/[EMAIL PROTECTED])


I dont know how to write the Dial parameters to say that I want to call 
number 150 of test
context in asterisk B server. So, I always fall into asterisk B default 
context.


Do anyone know how to write it?


Thanks you



Ngo Duc Loi a écrit :

dear miche,
 
pls place your number of softphone B into the context test dial plan.
 
with best regards,

osochebol

- Original Message 
From: Michel [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 9, 2007 9:44:20 AM
Subject: [asterisk-users] ooh323c calls

Hi,

I have two asterisk servers where softphone A is connected to asterisk A.
On those two asterisk servers,  ooh323c is installed.

I tried to call a test context on asterisk B from softphone A.  But I
always fall into context default of asterisk B.
( I don't know how to tell asterisk A extensions.conf to call asterisk B
test context)

Here are conf files on asterisk  A :

ooh323.conf

[softA]   ; softphone A uses this channel
type=user
context=test
ip=10.0.0.1
port=1720
disallow=all
allow=gsm
allow=ulaw

[mypeer1]
type=peer
ip=10.0.0.2
port=1720  


extensions.conf

[test]
exten = 15,1,Answer()
exten = 15,n,Playback(vm-hello)
exten = 15,n,Dial(OOH323/150/mypeer1);or  exten =
15,n,Dial(OOH323/[EMAIL PROTECTED])
exten = 15,n,Hangup()



May I  use  a gatekeeper? I learnt that ooh323c can act as gatekeeper,
but I didn't success to configure it (I have gatekeeper is not
responding error!). Can one of my server acts as
gatekeeper and gateway?

Do anyone success to configure gatekeeper with ooh323c ?


Thanks you for you help!





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[asterisk-users] Which H323 module for asterisk

[Michel]

Hello,

I need your advice about H323 and asterisk!  ;) Which one do you advice 
me to choose H323 (only gateway mode)? ooh323? ooh323c?


Which one is the best H323 module to use with asterisk? Which one did 
you choose and why?

What is your return on experience?



For more informations : http://www.voip-info.org/wiki-Asterisk+H323+channels



Thanks you for your replies!

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[asterisk-users] one way audio when forwarding from ser to asterisk

[richard Coco]

Hi all,

i have ser and asterisk on the same box with a public
ip address. When an UA behind NAT registred on SER try
to call the Voicemail or another UA registred on
Asterisk i have one way audio (caller cannot hear the
callee).

[UA/SER]--[router/nat]--[SER/Asterisk]


UA has private IP(192.168.204.19) and public IP is
89.106.xxx.yyy
SER/ASterisk has public ip (89.106.yyy.zzz).

In the sip trace one can see that signaling is ok but
Asterisk sends RTP from 89.106.xxx.zzz to
192.168.204.19 not to 89.106.xxx.yyy

ps: when UA registred on SER try to call UA2 registred
on SER every thing works fine.

how can i fix this issue.
thx   


 

Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
http://new.mail.yahoo.com
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[asterisk-users] DTMF on Snom

[asterisk]

Hi all,

I have problem using DTMF on Snom Phones (300, 320 and 360)

I read they use in preference out-of-band DTMF , and if the remote system
does not support it they default back to inband.

I would like to use DTMF as out of band , and I defined
dtmfmode=rfc2833
in the peer configuration.

Nope, I am no able to access any ouside services using DTMF;

Another kind of phones, ATCOM AT320, can be configured as inbond or out of
band;

Again, If I use out of band (rfc2833) I am not able to use DTMF
If I use
dtmfmode=inband in the peer definition and in the phone configuration, it
works ( I am using g711 codec)

Anyway I wouldn't  like to use inband, I would prefer to use gsm codec as I
know inband does not work very well with gsm codec

So the problem is: why my asterisk box defect on using rfc2833 ?
It is not a phone problem (4 phones of 2 different brands behave in the
same way)

What can I check in my asterisk configuration ?

thanks in advance,
Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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[asterisk-users] Sip dynamic host question

[Ale]

Hi all,

My asterisk box have some peers with as host the name of a dynamic dns 
resolver ex: foo.dyndns.org.


All works fine, until the host foo.dyndns.org for some reason change his 
ip, asterisk didn't resolve again the new ip until a sip relolad


Actually, i use a cron with a bash script to track the ip and eventually 
 reload the sip.conf.


Any tips for Asterisk ? Something like externrefresh for a peer?

Thanks,
Alessandro
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Re: [asterisk-users] Sip dynamic host question

[Vicky]

Asterisk can manage dynamic hostnames itseld type dnsmgr refresh in
asterisk cli  . Also see /etc/asterisk/dnsmgr.conf

On 10/01/07, Ale [EMAIL PROTECTED] wrote:


Hi all,

My asterisk box have some peers with as host the name of a dynamic dns
resolver ex: foo.dyndns.org.

All works fine, until the host foo.dyndns.org for some reason change his
ip, asterisk didn't resolve again the new ip until a sip relolad

Actually, i use a cron with a bash script to track the ip and eventually
  reload the sip.conf.

Any tips for Asterisk ? Something like externrefresh for a peer?

Thanks,
Alessandro
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Re: [asterisk-users] Snom side car annoyance

[Steve Davies]

Hi,

On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote:

Has anyone got this annoying sidecar to illuminate when users are on the
phone?


Yup, works fine.


I've tried Context: Line, still no dice. In extensions.conf I have:

exten = 4000,hint,SIP/4000,name


Make sure that the hint is not the first line referring to exten
4000. That seems to make a difference. Also, what is ,name doing at
the end of the line? I've never seen that done before.


Using Asterisk 1.2.13 on FC5, Snom:

Phone Type: snom360-SIP
Kernel Version: snom360 linux 3.25
Application-Version: snom360-SIP 6.5.2
Rootfs-Version: snom360 jffs2 v3.36


I have a similar setup.

Regards,
Steve
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Re: [asterisk-users] Which H323 module for asterisk

[Pavel Jezek]

I prefer h323 included in asterisk tree,
I have caller id issues with ooh323 and nobody answer to bugreports
oh323 from inaccessible network is unmaintained/death project, 
incompatible with asterisk 1.4.

PJ



Michel wrote:

Hello,

I need your advice about H323 and asterisk!  ;) Which one do you 
advice me to choose H323 (only gateway mode)? ooh323? ooh323c?


Which one is the best H323 module to use with asterisk? Which one did 
you choose and why?

What is your return on experience?



For more informations : 
http://www.voip-info.org/wiki-Asterisk+H323+channels




Thanks you for your replies!

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Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?

[Mark Coccimiglio]

M.Hockings wrote:

I don't really know the name of what I want to look for but maybe 
someone could tell me if it would be available.


I have a number of old analogue cell phones laying about here and I 
was thinking it would be useful if I could set up a short range base 
station for them that would cover maybe an acre or so.  What I would 
like to be able to do is use it to connect into Asterisk and this way 
have a useful wireless extension-phone range.


I do know that there are WiFi IP phones available but based on the 
connection range to our WiFi access points it seems limited as is our 
existing wireless handset (POTS).


Any thoughts, suggestions ?

Mike


You have a few options...

Firstly I would suggest throw away or donate the old phones.  There is 
much better technology then Analog Cellular.


Simple Choice 1:
  Get new GSM phones subscribed on the same carrier and a GSM 
terminal.  Make sure the phones all have free in-network calling 
(assuming that option is available in your country).  Also setup the GSM 
terminal on the same group and hook it up to your asterisk server (think 
of it as a cellular extension).  Lock the phones so that they can only 
call each other and the GSM terminal.


Cost: (assuming 5 phones  1 terminal) ~$2000 to start and 
$150-200/mo.  YMMV


More complex choice 2:
Get an RF engineer to design you a real WiFi coverage footprint and 
Wifi phones.


   Cost: $4000-7000 (or more) for consultation, hardware and setup.  No 
reoccuring charges (hopefully).







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[asterisk-users] libpri Calling Line ID

[Michael Konietzny]

Hey users,

i've got a question about calling line id in libpri - zaptel with 
switchtype q.sig. My Q.Sig partner is a Siemens F900 (HiPoint). If I 
enable

span debug i see messages from type CONNECT with some kind of bit field:

 Protocol Discriminator: Q.931 (8)  len=87
 Call Ref: len= 2 (reference 86/0x56) (Terminator)
 Message type: CONNECT (7)
 [1c 1d 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 0f 02 02 4b 36 02 01 55 
30 06 82 04 06 1c 08 40]
 Facility (len=31, codeset=0) [ 0x9f, 0xaa, 0x06, 0x80, 0x01, 0x00, 
0x82, 0x01, 0x00, 0x8b, 0x01, 0x00, 0xa1, 0x0f, 0x02, 0x02, 'K6', 0x02, 
0x01, 'U0', 0x06, 0x82, 0x04, 0x06, 0x1c, 0x08, 0x40 ]
 [1c 29 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 1b 02 02 4b 45 02 01 02 
a1 12 04 0d 4e 4f 52 44 4d 41 4e 4e 2c 45 52 49 43 02 01 01]
 Facility (len=43, codeset=0) [ 0x9f, 0xaa, 0x06, 0x80, 0x01, 0x00, 
0x82, 0x01, 0x00, 0x8b, 0x01, 0x00, 0xa1, 0x1b, 0x02, 0x02, 'KE', 0x02, 
0x01, 0x02, 0xa1, 0x12, 0x04, 0x0d, 'NORDMANN', 0x2c, 'ERIC', 0x02, 
0x01, 0x01 ]

 [4c 06 00 80 32 35 37 37]
 Connected Number (len= 8) [ Ext: 0  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0)
 Ext: 1 Presentation: Presentation 
permitted, user number not screened (0) '2577' ]



There is also a name included: NORDMANN ERIC. Is there any way in 
asterisk to get this name in a variable or by any applikation command ?


Thanks for your help in advance!

Cheers,
Michael Konietzny
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Re: [asterisk-users] Fax through Sangoma A102

[jeremij jerome]

Thank you all,

we succeeded to make the fax working synchronizing the clocks.

Regards,
Jeremi




On 1/9/07, Lee Howard [EMAIL PROTECTED] wrote:


jeremij jerome wrote:

 The problem is with the fax. We just want to send and receive faxes
 from/to our fax machine connected to the Siemens (without needing any
 interaction with our VoIP network, the faxes are sent to/received from
 PSTN). Unfortunately we are experiencing a lot of problems: the faxes
 not always work and when they work, it's likely to have incomplete
pages.


What are you using to fax?  Fax machines connected to ATAs?
txfax/rxfax?  IAXmodem and HylaFAX?

If you are using IAXmodem and HylaFAX a fax session log (HylaFAX) would
be quite revealing.

Lee.
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Re: [asterisk-users] getting tones during conversation

[Time Bandit]

after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status.  is this possible?


Have a look here :
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf

applicationmap is what you are looking for

hth
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[asterisk-users] Calls die when the answering party transfers

[Mohamed A. Gombolaty]

Dear All,

I am facing a strange problem that I can't find any matches for while
googling,  sometimes while a call initiated from asterisk to the PSTN is
answered and the answering party say the receiptionist tries to transfer
the call to someone else, the call dies, the full log shows nothing
useful and I am really unable to move forward on this issue, so can some
one suggest anything?

My zapata.conf is below also we are using Digium TDM400P with FXO
modules to connect to the PSTN.


[channels]
callerid = asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
busydetect=yes
immediate=no
faxdetect=both
busycount=4
callgroup=1
pickupgroup=1
pridialplan = local
prilocaldialplan = local
nationalprefix = 1
internationalprefix = 1011
group = 0
context=from-pstn
signalling=pri_cpe
switchtype = euroisdn
language=en
channel = 1-15,17-31

signalling=fxs_ks
context=from-zaptel
group=3
channel = 63-74


signalling=fxs_ks
context=from-zaptel
group=4
channel = 75-78

--
Thx
MAG


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Re: Re: [asterisk-users] Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'

[phil . dawson]

This page should help:

http://www.voip-info.org/wiki/view/Asterisk+CentOS-4.0+Zaptel






   
 Tzafrir Cohen 
 [EMAIL PROTECTED] 
 rcom.com  To 
 Sent by:  asterisk-users@lists.digium.com 
 asterisk-users-bo  cc 
 [EMAIL PROTECTED] 
 m.com Subject 
   [SPAM] Re: [asterisk-users] Zap 1.4 
   error line 0: Unable to open master 
 10/01/2007 06:23  device '/dev/zap/ctl'   
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




On Tue, Jan 09, 2007 at 05:50:52PM -0600, Chris Bullock wrote:
 I've looked over EVERY resource I can find, but have run short of a
 solution.  I'm running CentOS 4.4.  Just installed Asterisk 1.4 and
Zaptel
 1.4 and libpri, but when I run ztcfg I get this error: line 0: Unable to
 open master device '/dev/zap/ctl'

This is a generic error message ztcfg gives when it fails to open
/dev/zap/ctl. It is followed by the error string of the error code it
got (usually: no such file or no such device).

No such file: the file /dev/zap/ctl is simply not there.

No such device: The file is there, but there is no device to support
it.

If you use udev (or the older devfs) and have not created the device
file yourself manually with mknod, you probably won't get the latter.


 I realize this is a udev error (or from what I've read), but I cannot
find
 out how to resolve this. I've reinstalled zaptel several times. I read a
lot
 about having to read the README.udev file in the zaptel source, but I
don't
 even have that file on my system.

 If anyone has any ideas I'd love to hear from them.

It may be because the module zaptel has failed to load. Do you have the
directory /proc/zaptel ?

lsmod | grep zaptel

--
   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] Zap 1.4 error line 0: Unable to open

[Chris Bullock]

Here is the complete output of ztcfg:

Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected



It appears that none of the zaptel devices have been created. I did not
notice any errors during the make install.  Does anyone have any
suggestions?  I know the hardware works, because it was working as my
asterisk 1.2 test system before I reloaded it completely and installed
asterisk 1.4

I appreciate the help.

-Chris

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Re: [asterisk-users] Zap 1.4 error line 0: Unable to open

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 08:03:07AM -0600, Chris Bullock wrote:
 Here is the complete output of ztcfg:
 
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'
 
 1 error(s) detected
 
 
 
 It appears that none of the zaptel devices have been created. I did not
 notice any errors during the make install.  Does anyone have any
 suggestions?  I know the hardware works, because it was working as my
 asterisk 1.2 test system before I reloaded it completely and installed
 asterisk 1.4
 
 I appreciate the help.

/me repeats the followup-question:

It may be because the module zaptel has failed to load. Do you have the 
directory /proc/zaptel ?

lsmod | grep zaptel 

  

-- 
   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] Where is this hilarious Allison Smith file? (Also: Interview with Allison)

[Kevin P. Fleming]

Jerry Glomph Black wrote:
 I cannot find this file anywhere, despite thorough searching.
 Certainly not in the two usual big sound tarfiles.   I have a great
 place for this file in my extensions.conf, no doubt.

It has not been made available for distribution, sorry.
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Re: [asterisk-users] Problem with zaptel drivers or card

[Matthew Fredrickson]

On Jan 9, 2007, at 7:01 PM, Administrator wrote:

I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 
1.4, and Zaptel 1.4

The Digium cards installed are TDM2400 and TE110P.
Everything was working fine until I upgraded to zaptel 1.2.12 from 
1.2.9

Now when I run ztcfg I get the following error message:
(CAS signalling on span 2 conflicts with Clear channel on channel 40)
--NOTE: signaling was spelled wrong in the error message--
I have since upgraded to 1.4 with the same problem.
Channel 40 is a standard bchan configuration and our provider sees no 
problem with the channel.

When I disable the channel everything works fine.
My assumption is that something is wrong with the TE110P card.
Has anyone seen anything else like this?


What's in your zaptel.conf and can you post the output of `cat 
/proc/zaptel/1` and `cat /proc/zaptel/2`?


Matthew Fredrickson

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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Ed Rubright - mail lists]

Mark Coccimiglio wrote:

Marty,
   Where are you paying $1000 for a 1600 series Cisco?  I can get you 
20% off that price on any quantity (note: Sarcasam).  Its not the 
1990's anymore.  You can get them on eBay ($50-150) for only slightly 
more then the Linksys.  The performance is rock solid.  Three-quarters 
of the world have used them for decades.  I know of units running 2 
and 3 YEARS between reboots.  The power company reboots my equipment 
more then I do.  Ok it is true that Cisco does not support the models 
anymore, but you can't buy a services contract for a linksys router 
either.  It can sometimes be a little difficult to configure without 
any technical knowledge but that is what most of us get paid for.  It 
does impress the customer when you bring in the grey box labled 
Cisco.  As for performance just try to put 50 people behind a 
linksys/netgear/dlink.  I've used 1605R supporting +100 users.  Not 
even a blink.  Finally, untill everyone is using 10Mps FTTH the 
broad band link is still the slowest part of the connection.  Not to 
shabby for antiquated technology.


Mark C

Martin Joseph wrote:


On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:


Mike
I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router 
with Fair-Weight queueing enabled.  Works great.  The nice thing 
about Fair-Weight queueing is that it dynamically adapts to lower 
the priority of higher demand traffic (e.g. large downloads).  If 
you want quality stick with quality stuff.


Mark C



Reread the subject line please.  $1000 (US) isn't inexpensive by any 
stretch.


Marty


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Mark,

Do these 1600 series Cisco routers you mention that you find on eBay for 
$50-$150 support Layer3 routing?  I have a managed switch setup on my 
home network with several VLANs defined. (work subnet, home subnet, VOIP 
subnet)   I currently have to use a Linux box to route between the 
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the 
Linux box(more processor power and new NICs) and that gets expensive.


I'd much rather have a router or smart switch for that matter that does 
Gigabit Layer3 routing all in one unit. 


Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed
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[asterisk-users] SPA-3000 and Asterisk 1.4.0

[Thomas Kenyon]

Has anyone else had any difficulty with calls Originating from the PSTN 
being passed to asterisk 1.4.0 unsing a linksys SPA-3000?


I've not had enough time to track down what's happening but with 1.4.0, 
When a call comes in, it is passed to asterisk and then forwarded to the 
 extension that rings, but when the extension is lifted the call hangs up.


This does not happen with 1.4.0b2 (which I have rolled back to for now, 
but will try again soon when I get some free downtime).

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[asterisk-users] Asterisk HA

[Enrico Pasqualotto]

Hi all, I have to make for a client an asterisk system for process up to 
250 calls between conference and normal call.
At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client 
require a failover system.

Anyone have experience for this type of solution?
Is better ultramonkey, dundi or SER proxy in front of * server?

Thanks  Enrico

P.S. Now during all this year I have to work with this type of solution, 
 why not make a fork of this ml  for example 
[EMAIL PROTECTED], for write some docs too.

--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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[asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Michael Hamann]

Hello,

we are running a Asterisk (1.2) installation with about 80 snom phones
(300,320,360).

Now have the demand for a special manager - assistant setup for a few
extensions.

Since Shared Line Appearance is not available in 1.2 I´m wondering how
to realize this...

What we need is that the manager can decide whether he wants to get
calls or not. If not he must have the possibility to redirect all
incoming calls to his secretary. The secretary itself answers all calls
and decides if the call is important enough to disturb the manager. If
so she/he transfers the call to the manager. So the secretary can filter
the calls for the manager...

The only way I can imagine so far is via a redirect by AstDB on the
manager extension. The managers phone has two different lines - the
official and a secret one only the secretary uses...

Or are there any other solutions?

Any hint will be appreciated ...

Michael
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Re: [asterisk-users] Attended Transfer on queue_log

[equis software]

Yes, I have de same problem...I dont know if there is an error...

Regards

On 12/15/06, Miguel Paolino [EMAIL PROTECTED] wrote:


I'm using asterisk blind/attended transfer feature on  a queue (also
tried with sip phones feature), and both type of transfers work fine. The
problem is that attended trasfers doesn't get logged to queue_log, but blind
transfers are logged just fine. Anyone knows if this is the correct
behavior?

--
Regards,

Miguel Paolino
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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Gary Richardson]

I'd wager to say yes, it does support layer 3 routing :) That's a bit of a
redundant term (though you can route above layer 3). Depending on how many
interfaces you have on your router, you may be sending multiple vlans over a
trunk port (I'm pretty sure the 1600 series support trunk ports -- you may
want to google 'router on a stick').

Most of the layer 3 gigabit switches will still be very expensive, though
Catalyst 3500's may be getting 'cheaper' -- most of the 3500 and 3700 series
switch have multi-gigabit backplanes (usually 16 - 32 gigabits) and can
usually route packets are wire speed, or very close to it. If you are
looking for a gigabit port or two for uplink, I believe they even made a
2900G, though that won't have PoE. And now that I think about it, probably
doesn't support layer 3 routing :(

That's the Cisco world, I'm sure you can find other vendors that have
hardware for much cheaper, though this is an advantage to using the same
networking equipment most other people are using. Also, most of this is
overkill for a handful of network devices.


On 1/10/07, Ed Rubright - mail lists [EMAIL PROTECTED] wrote:



Do these 1600 series Cisco routers you mention that you find on eBay for
$50-$150 support Layer3 routing?  I have a managed switch setup on my
home network with several VLANs defined. (work subnet, home subnet, VOIP
subnet)   I currently have to use a Linux box to route between the
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the
Linux box(more processor power and new NICs) and that gets expensive.

I'd much rather have a router or smart switch for that matter that does
Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed
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[asterisk-users] Send email notification

[H Aranguren]

Hi group,

I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).

I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo Hello | sendmail [EMAIL PROTECTED] -f [EMAIL 
PROTECTED]).


My voicemail.conf
[general]
format=wav49
attach=yes
[EMAIL PROTECTED]
fromstring=Asterisk
mailcmd=/usr/sbin/sendmail -t
[my_home]
100 = ,number100,[EMAIL PROTECTED]

My sip.conf
[100]
type=friend
secret=pass
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
[EMAIL PROTECTED]


Can you see the problem?. Do you know any documentarion on internet
where can i solve the problem?

Regards,
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[asterisk-users] RTP directly

[David Alcott]

Is there a way to configure the Asterisk so that the RTP goes directly 
between the Endpoints as opposed to going through the asterisk?


-Dave


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[asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

Hi,

I'm trying to write a AGI in PHP to get the numbers dialed (with read()),
save it into a variable to insert it into a SQL server database. But I
cannot see results into the variable, it always return NULL.
Here is a piece of the AGI.

fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));

Even if I only show my_var value or try to use it inside asterisk, the value
is NULL.
There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.

Thank you all.

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Henry.L.Coleman]

Hi Michael, in practice I think that the managers extension should default
to the assistant who can screen the call or call forward it.
Call Forward - always or Call Forward - no answer would give you the
flexability required.


Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 Hello,

 we are running a Asterisk (1.2) installation with about 80 snom phones
 (300,320,360).

 Now have the demand for a special manager - assistant setup for a few
 extensions.

 Since Shared Line Appearance is not available in 1.2 I´m wondering how
 to realize this...

 What we need is that the manager can decide whether he wants to get
 calls or not. If not he must have the possibility to redirect all
 incoming calls to his secretary. The secretary itself answers all calls
 and decides if the call is important enough to disturb the manager. If
 so she/he transfers the call to the manager. So the secretary can filter
 the calls for the manager...

 The only way I can imagine so far is via a redirect by AstDB on the
 manager extension. The managers phone has two different lines - the
 official and a secret one only the secretary uses...

 Or are there any other solutions?

 Any hint will be appreciated ...

 Michael
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Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

[Christoph Adomeit]

 Option A: Use the manager interface.
 
Tzafrir , Thanks,

the idea to use the manager interface is wonderful. It is really fast
and no data gets lost. I don't think 4000 Rows are a noticeable 
amaount of data for a db1 database.

I coded this:
#!/usr/bin/perl

use Asterisk::Manager;

my $astman = new Asterisk::Manager;

$astman-user('admin');
$astman-secret('bla');
$astman-host('localhost');
$astman-connect || die Could not connect to  . $astman-host . !\n;

foreach $num(1..5000) {
  $astman-command(database put callerids willi$num $num);
}

$astman-disconnect;



-- 
Two hours of trial and error can save ten minutes of manual reading.
GATWORKS GmbH
[EMAIL PROTECTED] Internetloesungen vom Feinsten
Fon. +49 2166 9149-32  Fax. +49 2166 9149-10
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Thanks for the help. I was concerned because I tried once before and it
formatted my hard disk. I wanted to be sure that did not happen again.\
Bob Rawlinson

On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
 Has anyone heard of a build or instructions for installing Asterisk on a
 Suse 10.1 system?
 Bob Rawlinson
 
 
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[asterisk-users] Directory too difficult?

[Colin Anderson]

I have a group of users whos complaint about Asterisk is that the directory
application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to create a
custom directory for these guys. Anyone have any tips for making the
directory easier, maybe re-record the prompts so they are more verbose? We
go by first name. 
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Re: [asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
 Hello,
 
 we are running a Asterisk (1.2) installation with about 80 snom phones
 (300,320,360).
 
 Now have the demand for a special manager - assistant setup for a few
 extensions.
 
 Since Shared Line Appearance is not available in 1.2 I´m wondering how
 to realize this...
 
 What we need is that the manager can decide whether he wants to get
 calls or not. If not he must have the possibility to redirect all
 incoming calls to his secretary. The secretary itself answers all calls
 and decides if the call is important enough to disturb the manager. If
 so she/he transfers the call to the manager. So the secretary can filter
 the calls for the manager...
 
 The only way I can imagine so far is via a redirect by AstDB on the
 manager extension. The managers phone has two different lines - the
 official and a secret one only the secretary uses...
 
 Or are there any other solutions?
 
 Any hint will be appreciated ...

Hello Michael,

as I see it, the most obvious setup would be

- have SIP accounts, e.g. sip123 for the secretary phone, sip456 and
sip789 for the manager phone.
- the official/public extension number for the manager might be
4321, so

exten = 4321,1,Dial(SIP/sip123SIP/sip456)

would ring both the secretary phone and the manager phone on the
public id (which most probably can have a separate ringtone than the
private id). You would also want a private extension like

exten = 4901,1,Dial(SIP/sip789)

for the secretary to reach the manager. A few thoughts:
- The Callerid setting for both secretary and chief should be 4321, no
matter which line the chief chooses to call out through.
- Do not choose an obvious private number, like 4321 and 4322
- You could even choose a real long number, that only is available
from internal phones, and put it to a speed dial button on the secretary
phone

If you want the manager to be able to selectively not be disturbed by
public number calls, but only by his secretary, some AstDB logic could
come into the game. This can be highly dynamic, or you just configure a
few extensions by hand to do exactly this:

exten = 770/4321,1,Set(DB(list/4321)=SIP/sip123SIP/sip456)
exten = 770/4321,2,Playback(feature-donotdisturb-off)
exten = 771/4321,1,Set(DB(list/4321)=SIP/sip123)
exten = 771/4321,2,Playback(feature-donotdisturb-on)
exten = 4321,1,Dial(${DB(list/4321)})

So either the chief or the secretary could activate do-not-disturb by
dialing 771, and deactivate with 770. Just examples; choose those codes
from a range that is not in use as extensions; for my personal setup,
the 2*/3*/4*/5*/6* internal numbering for SIP devices, OOH devices, IAX
devices etc.pp., 8* being applications (like 888 the talking clock), 9*
experimental and 0* PSTN calls (how 80's! :-). A somehow similar
function (divert to VoiceMail delay in seconds can be set from any
phone, between 0 and 60 seconds) is available here as 811x.
Choose whatever suits you best.

Of course one could imagine also that the manager phone number NOT rings
the secretary while the manager is there and ready to take calls - just
edit the 770/771 lines (or add 772 for that function) - in that case,
the secretary could make use of an extension number for him/herself, but
her phone also has several lines, so why not.

HTHBR
Anselm

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[asterisk-users] Service Level Compliance

[lists]

Hello all,

We have a slight issue to resolve. We have a client who we are drafting an SLA 
for the delivery of telephony services using Asterisk. Nothing extraordinary. 
However, we do need a way to measure our service availability.

We currently use Nagios and Cacti to monitor server availability as well as 
asterisk and mysql responsiveness, and last, ping availability to our 
origination VoIP providers. In an ideal world, this should be fine. However, 
there are a few cases we have noticed this setup not to be enough.

Our particular setup is origination traffic comes into Asterisk box A, where 
the call goes through some AGI-based IVR. After navigating thru the IVR, the 
call is transfered to Asterisk box B, where the call is put in a queue and 
distributed to SIP-based agents.

The issues we would like to resolve are the following:

1) We can ping our originating SIP providers. However, that doesn't guaratee us 
that we can receive calls from them. In several occasions, some of our SIP 
providers have had routing (SIP) problems and when we dial any of the DIDs, 
they would not even hit our box. The call would simply die somewhere in their 
network or their providers' networks. How can we proactively confirm that they 
are actually routing calls to us? We thought we could probably dial out through 
any of our other providers so the call comes in via a different provider and 
maybe hit an AGI script. This script could update a MySQL table with a 
timestamp of the last successful test. We could then take the data from that 
table and bring it to Nagios and/or Cacti. Is there a better approach?

2) We can test Asterisk responsiveness by doing something like 'asterisk -rx 
show uptime' and parse the results. We can also connect to MySQL and execute 
a test query. However, how can we verify that Asterisk is actually talking to 
MySQL and that it's connection hasn't died?

3) As stated above, we can test the responsiveness of asterisk. However, we 
have noticed in, at least, one occasion, that even though asterisk seems to be 
responsive, it would not accept or place any calls. Somehow it's call engine 
was locked and we had to restart asterisk. How can we verify that asterisk is 
actually capable of receiving and placing calls?

4) We have no Digium boards and all kernels are 2.6 or above, so we end us 
using ztdummy, if needed. The client's agents are in a different country and 
the average lantency is around 250ms. Most of the time, call quality is good. 
However, there are a few situations where people complaint about echo. Is there 
a way to measure or improve this? I know it has been a topic discussed at 
lenght and if we could probably script a way to measure some sort of a MOS 
value, that would be great. Any ideas?

5) Anything else that you could think of we could measure to make sure all 
components are working?

You input is greatly appreciated it. I promise that whatever solution is best 
recommended and scriptable, we will post our development and working solutions 
for the community to benefit from.

Thanks again,
Daniel

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Re: [asterisk-users] Send email notification

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:
 Hi group,
 
 I'm trying to configure the email notification when a user leave a
 voicemail, but don't work (send email notification).
 
 I configured esmtp in my linux box, if a try to use it with command
 line, it works fine. (echo Hello | sendmail [EMAIL PROTECTED] -f [EMAIL 
 PROTECTED]).

You could look wether a voicemail triggers something to happen inside
the mail system at all (logfiles...). In that case, chances are that the
mail cannot be parsed because of misconfiguration in the mail server /
restricted usage of the sendmail -t command or whatever.
In my setup (SMTP server listening on port 25 of the same machine) the
mailcmd is commented out, and It Just Works(tm). If you need mail system
specific help, there sure are lots of forums and info, but I cannot tell
where to connect to esmtp people. Exim is my favourite ;)

 
 My voicemail.conf
 [general]
 format=wav49
 attach=yes
 [EMAIL PROTECTED]
 fromstring=Asterisk
 mailcmd=/usr/sbin/sendmail -t
 [my_home]
 100 = ,number100,[EMAIL PROTECTED]
 
 Can you see the problem?. Do you know any documentarion on internet
 where can i solve the problem?

BR
Anselm

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Re: [asterisk-users] Get dialed numbers in AGI

[Anton Frolov]

you could use one of the AGI libraries...
then you can just call a function to get the number.

AF.


Ralph Liebessohn wrote:
 Hi,
 
 I'm trying to write a AGI in PHP to get the numbers dialed (with
 read()), save it into a variable to insert it into a SQL server
 database. But I cannot see results into the variable, it always return
 NULL.
 Here is a piece of the AGI.
 
 fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
 fflush(STDOUT);
 $conn=odbc_connect('MSSQL', 'USER', 'PASS');
 $query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));
 
 Even if I only show my_var value or try to use it inside asterisk, the
 value is NULL.
 There is another way to do it? Am I doing a mistake here?
 I'm using Asterisk 1.2.13.
 
 Thank you all.
 
 -- 
 Ralph Liebessohn
 ICQ: 74835911
 Skype: liebessohn
 
 
 
 
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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

Results From cat /proc/zaptel/*

Span 1: WCTDM/0 Wildcard TDM2400P Board 1
IRQ misses: 24

   1 WCTDM/0/0 FXOKS (In use)
   2 WCTDM/0/1 FXOKS (In use)
   3 WCTDM/0/2 FXOKS (In use)
   4 WCTDM/0/3 FXOKS (In use)
   5 WCTDM/0/4 FXOKS (In use)
   6 WCTDM/0/5 FXOKS (In use)
   7 WCTDM/0/6 FXOKS (In use)
   8 WCTDM/0/7 FXOKS (In use)
   9 WCTDM/0/8 FXOKS (In use)
  10 WCTDM/0/9 FXOKS (In use)
  11 WCTDM/0/10 FXOKS (In use)
  12 WCTDM/0/11 FXOKS (In use)
  13 WCTDM/0/12 FXOKS (In use)
  14 WCTDM/0/13 FXOKS (In use)
  15 WCTDM/0/14 FXOKS (In use)
  16 WCTDM/0/15 FXOKS (In use)
  17 WCTDM/0/16 FXOKS (In use)
  18 WCTDM/0/17 FXOKS (In use)
  19 WCTDM/0/18 FXOKS (In use)
  20 WCTDM/0/19 FXOKS (In use)
  21 WCTDM/0/20 FXOKS (In use)
  22 WCTDM/0/21 FXOKS (In use)
  23 WCTDM/0/22 FXOKS (In use)
  24 WCTDM/0/23 FXOKS (In use)
Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 B8ZS/ESF

  25 WCT1/0/1 Clear (In use)
  26 WCT1/0/2 Clear (In use)
  27 WCT1/0/3 Clear (In use)
  28 WCT1/0/4 Clear (In use)
  29 WCT1/0/5 Clear (In use)
  30 WCT1/0/6 Clear (In use)
  31 WCT1/0/7 Clear (In use)
  32 WCT1/0/8 Clear (In use)
  33 WCT1/0/9 Clear (In use)
  34 WCT1/0/10 Clear (In use)
  35 WCT1/0/11 Clear (In use)
  36 WCT1/0/12 Clear (In use)
  37 WCT1/0/13 Clear (In use)
  38 WCT1/0/14 Clear (In use)
  39 WCT1/0/15 Clear (In use)
  40 WCT1/0/16
  41 WCT1/0/17 Clear (In use)
  42 WCT1/0/18 Clear (In use)
  43 WCT1/0/19 Clear (In use)
  44 WCT1/0/20 Clear (In use)
  45 WCT1/0/21 Clear (In use)
  46 WCT1/0/22 Clear (In use)
  47 WCT1/0/23 Clear (In use)
  48 WCT1/0/24 HDLCFCS (In use)

zaptel.conf file:

fxoks=1-24
span=2,1,0,esf,b8zs
bchan=25-39
#bchan=40
bchan=41-47
dchan=48
loadzone=us
defaultzone=us

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, January 09, 2007 11:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
 I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
and
 Zaptel 1.4
 
 The Digium cards installed are TDM2400 and TE110P.
 
 Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
 
 Now when I run ztcfg I get the following error message:
 
 (CAS signalling on span 2 conflicts with Clear channel on channel 40)
 
 --NOTE: signaling was spelled wrong in the error message--
 
 I have since upgraded to 1.4 with the same problem.
 
 Channel 40 is a standard bchan configuration and our provider sees no
 problem with the channel.
 
 When I disable the channel everything works fine.
 
 My assumption is that something is wrong with the TE110P card.
 
 Has anyone seen anything else like this?

What do you get from:

cat /proc/zaptel/*

What do you have on /etc/zaptel.conf  ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] 1.4 and zap bugs

[Julian Lyndon-Smith]

We're currently running 1.4 r48326 - a little while before the full 1.4 
release.


We are having some problems (crashes) with attended transfers and some 
other stuff, and was going to move to the latest svn 1.4 as I beleive 
that the attended transfer bug has been fixed.


However, I note that #8763 (http://bugs.digium.com/view.php?id=8763) has 
some problems with the zap channels in the 1.4.0 release (which we 
*don't* have).


My question is, are the problems with zap also theoretically present in 
r48326 (it's just that we don't have them) or were they introduced after 
r48326 (and therefore we will have them if we upgrade).


Any thoughts / takers / advice ?

Julian.
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Doug Crompton]

Formated your hardisk... wow that is nasty, but I also cannot understand
how that could ever happen. There must be some other HW problem going on
or you got a hold of some really bad code.

What code (source or binary) and what were you doing when that happenned?

Doug

On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:

 Thanks for the help. I was concerned because I tried once before and it
 formatted my hard disk. I wanted to be sure that did not happen again.\
 Bob Rawlinson

 On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
  Has anyone heard of a build or instructions for installing Asterisk on a
  Suse 10.1 system?
  Bob Rawlinson
 
 
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k

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[asterisk-users] app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'

[Chris Bullock]

When I load the asterisk 1.4 gui and log into
/asterisk/static/config/setup/install.html, it tells me No Analog ports
has been detected on your system.

I have 2 Wildcard X100P cards that are properly installed. Ztcfg shows no
problems.

I also get the following message from the asterisk console 
app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'
when I log into the web interface.

Any ideas?

-Chris

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Re: [asterisk-users] Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'

[Chris Bullock]

Ok. I finally got past this. After doing all the relevant udev stuff, I ran
a make config from the zaptel sources, and got the service to install.

I'm still quiet an asterisk newbie, and defiantly a huge Linux newbie, so
thanks for the help.

-Chris

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Re: [asterisk-users] Directory too difficult?

[Bryan M. Johns]

I wish had some pearl of wisdom here, but I don't.  I will simply  
share my sympathy.


Sounds like an ESU situation to me.

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:

I have a group of users whos complaint about Asterisk is that the  
directory

application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to  
create a

custom directory for these guys. Anyone have any tips for making the
directory easier, maybe re-record the prompts so they are more  
verbose? We

go by first name.
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RE: [asterisk-users] Which H323 module for asterisk

[Dan Austin]

Pavel wrote:
 I prefer h323 included in asterisk tree,
 I have caller id issues with ooh323 and nobody
 answer to bugreports oh323 from inaccessible 
 network is unmaintained/death project, incompatible
 with asterisk 1.4.
 PJ
Response to ooh323c bugs is very slow, and patches can
take some time to be applied if you manage to fix the
issue for yourself.

That said I prefer ooh323c, as it does not require
OpenH323 or PWlib.  I find building it easier.


 Michel wrote:
 Hello,

 I need your advice about H323 and asterisk!  ;) 
 Which one do you advice me to choose H323 
 (only gateway mode)? ooh323? ooh323c?
Since you mention gateway mode, then ooh323c is 
worth testing.  The bugs that I am aware of are
mostly gatekeeper related (but not all).  Since
the channel doesn't have any external dependencies,
it is the easiest to test.  If it doesn't work for
your setup, there's a very good chance that 
chan_h323 included with Asterisk will and then you
can deal with getting the OpenH323 and PWlib
dependencies meet. (Not a major issue, but one I
have preferred to avoid)


 Which one is the best H323 module to use with 
 asterisk? Which one did you choose and why?
 What is your return on experience?
Bugs happen.  I've found that the code for 
chan_ooh323c is reasonably easy to read and
make patches for.  The current release seems
stable and I have it running on four light to
moderately loaded servers.

Dan
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[asterisk-users] Proper use of the Local channel

[lists]

Is there any documentation you guys can point us to in order to learn more 
about the proper use of the Local channel? We don't currently use it. However, 
while evaluating other people's billing and management systems for Asterisk, we 
noticed they make extensive use of it.

Thanks,
Daniel

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Re: [asterisk-users] Problem with zaptel drivers or card

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 09:39:47AM -0700, Administrator wrote:
 Results From cat /proc/zaptel/*
 
 Span 1: WCTDM/0 Wildcard TDM2400P Board 1
 IRQ misses: 24
 
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXOKS (In use)
4 WCTDM/0/3 FXOKS (In use)
5 WCTDM/0/4 FXOKS (In use)
6 WCTDM/0/5 FXOKS (In use)
7 WCTDM/0/6 FXOKS (In use)
8 WCTDM/0/7 FXOKS (In use)
9 WCTDM/0/8 FXOKS (In use)
   10 WCTDM/0/9 FXOKS (In use)
   11 WCTDM/0/10 FXOKS (In use)
   12 WCTDM/0/11 FXOKS (In use)
   13 WCTDM/0/12 FXOKS (In use)
   14 WCTDM/0/13 FXOKS (In use)
   15 WCTDM/0/14 FXOKS (In use)
   16 WCTDM/0/15 FXOKS (In use)
   17 WCTDM/0/16 FXOKS (In use)
   18 WCTDM/0/17 FXOKS (In use)
   19 WCTDM/0/18 FXOKS (In use)
   20 WCTDM/0/19 FXOKS (In use)
   21 WCTDM/0/20 FXOKS (In use)
   22 WCTDM/0/21 FXOKS (In use)
   23 WCTDM/0/22 FXOKS (In use)
   24 WCTDM/0/23 FXOKS (In use)
 Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 B8ZS/ESF
 
   25 WCT1/0/1 Clear (In use)
   26 WCT1/0/2 Clear (In use)
   27 WCT1/0/3 Clear (In use)
   28 WCT1/0/4 Clear (In use)
   29 WCT1/0/5 Clear (In use)
   30 WCT1/0/6 Clear (In use)
   31 WCT1/0/7 Clear (In use)
   32 WCT1/0/8 Clear (In use)
   33 WCT1/0/9 Clear (In use)
   34 WCT1/0/10 Clear (In use)
   35 WCT1/0/11 Clear (In use)
   36 WCT1/0/12 Clear (In use)
   37 WCT1/0/13 Clear (In use)
   38 WCT1/0/14 Clear (In use)
   39 WCT1/0/15 Clear (In use)
   40 WCT1/0/16
   41 WCT1/0/17 Clear (In use)
   42 WCT1/0/18 Clear (In use)
   43 WCT1/0/19 Clear (In use)
   44 WCT1/0/20 Clear (In use)
   45 WCT1/0/21 Clear (In use)
   46 WCT1/0/22 Clear (In use)
   47 WCT1/0/23 Clear (In use)
   48 WCT1/0/24 HDLCFCS (In use)

Is it supposed to be T1 or E1?

The card behaves as E1 but you attempt to configure it as T1.

 
 zaptel.conf file:
 
 fxoks=1-24
 span=2,1,0,esf,b8zs
 bchan=25-39
 #bchan=40
 bchan=41-47
 dchan=48
 loadzone=us
 defaultzone=us
 
 James
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
 Sent: Tuesday, January 09, 2007 11:25 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Problem with zaptel drivers or card
 
 On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
  I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
 and
  Zaptel 1.4
  
  The Digium cards installed are TDM2400 and TE110P.
  
  Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
  
  Now when I run ztcfg I get the following error message:
  
  (CAS signalling on span 2 conflicts with Clear channel on channel 40)
  
  --NOTE: signaling was spelled wrong in the error message--
  
  I have since upgraded to 1.4 with the same problem.
  
  Channel 40 is a standard bchan configuration and our provider sees no
  problem with the channel.
  
  When I disable the channel everything works fine.
  
  My assumption is that something is wrong with the TE110P card.
  
  Has anyone seen anything else like this?
 
 What do you get from:
 
 cat /proc/zaptel/*
 
 What do you have on /etc/zaptel.conf  ?
 
 -- 
Tzafrir Cohen   
 icq#16849755jabber:[EMAIL PROTECTED]
 +972-50-7952406   mailto:[EMAIL PROTECTED]   
 http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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-- 
   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] Asterisk build for Suse 10.1

[Anton Frolov]

he is probably tried to install one of these All in one Asterisk CDs, that
effectively formats the hard drive and installs everything from scratch,
including the OS ;)

And, yes, it will happen again, if he re-runs this CD...

AF.


Doug Crompton wrote:
 Formated your hardisk... wow that is nasty, but I also cannot understand
 how that could ever happen. There must be some other HW problem going on
 or you got a hold of some really bad code.
 
 What code (source or binary) and what were you doing when that happenned?
 
 Doug
 
 On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:
 
 Thanks for the help. I was concerned because I tried once before and it
 formatted my hard disk. I wanted to be sure that did not happen again.\
 Bob Rawlinson

 On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
 Has anyone heard of a build or instructions for installing Asterisk on a
 Suse 10.1 system?
 Bob Rawlinson


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[asterisk-users] VIA EPIA DeadLock Issues

[Raymond McKay]

Greetings,

I've been having a large number of deadlock issues lately on chan_sip occurring 
only on VIA EPIA ML6000 boards.  I'm curious if anyone else is having similar 
issues.

My Config (have multiple systems all running the same hardware with the same 
problem)

VIA EPIA ML6000
1GB RAM
80GB HDD
Various Digium Cards (T1 and TDM cards)
Trixbox 1.2.2 (though running stock asterisk code)
Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch

Problem seems to happen more on systems that use parking lots.  The system will 
run for around 24 hours or so fine, and then mysteriously, without any errors 
leading up to it,  will stop being able to send calls to the chan_sip.  System 
from that point on reports the following in the logs.

Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook
Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait(Zap/1-1, 1) 
in new stack
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for 
'0x9896848', 10 retries!
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for 
'0x9896848', 10 retries!

attempting to stop asterisk from the CLI causes the CLI to become unresponsive 
and a trace shows chan_sip goes into a mutex_wait state. 

Anybody seen this? Have a fix?

Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
(860) 693-2226 x 31
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Re: [asterisk-users] Get dialed numbers in AGI

[Yuan LIU]

From: Anton Frolov [EMAIL PROTECTED]

you could use one of the AGI libraries...
then you can just call a function to get the number.

AF.

Ralph Liebessohn wrote:
 Hi,

 I'm trying to write a AGI in PHP to get the numbers dialed (with
 read()), save it into a variable to insert it into a SQL server
 database. But I cannot see results into the variable, it always return
 NULL.
 Here is a piece of the AGI.

 fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
 fflush(STDOUT);
 $conn=odbc_connect('MSSQL', 'USER', 'PASS');
 $query = odbc_exec($conn, INSERT INTO dialed(number) 
VALUES('$my_var'));


 Even if I only show my_var value or try to use it inside asterisk, the
 value is NULL.


Then there must be an error somewhere.  The variable READ() in Asterisk 
should be usable.  Should be able to use SayDigits() to play it back - or no 
value is read.


Yuan Liu


 There is another way to do it? Am I doing a mistake here?
 I'm using Asterisk 1.2.13.

 Thank you all.

 --
 Ralph Liebessohn
 ICQ: 74835911
 Skype: liebessohn



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[asterisk-users] Send email notification

[H Aranguren]

Thanks for your answer Anselm,

  But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the notification.

I used ethereal and i couldn't see any message from asterisk box to my
smtp server when i leave a voicemail

Thanks



Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:

Hi group,

I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).

I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo Hello | sendmail a at b.com -f b at c.com).


You could look wether a voicemail triggers something to happen inside
the mail system at all (logfiles...). In that case, chances are that the
mail cannot be parsed because of misconfiguration in the mail server /
restricted usage of the sendmail -t command or whatever.
In my setup (SMTP server listening on port 25 of the same machine) the
mailcmd is commented out, and It Just Works(tm). If you need mail system
specific help, there sure are lots of forums and info, but I cannot tell
where to connect to esmtp people. Exim is my favourite ;)



My voicemail.conf
[general]
format=wav49
attach=yes
serveremail=anonymous at abc.com
fromstring=Asterisk
mailcmd=/usr/sbin/sendmail -t
[my_home]
100 = ,number100,number100 at abc.com

Can you see the problem?. Do you know any documentarion on internet
where can i solve the problem?



BR
Anselm

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[asterisk-users] dundi ENCREJ

[Ramon Schönborn]

hi list,

i have the same problem as mentioned here:
http://forums.digium.com/viewtopic.php?t=2678view=nextsid=bd94cefd823b23156c5748843febb3ab

my asterisk version is 1.2.12.1

any ideas?






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RE: [asterisk-users] Directory too difficult?

[Alexander Lopez]

More like a ID-10-T error.

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?

 

I wish had some pearl of wisdom here, but I don't.  I will simply share
my sympathy.

 

Sounds like an ESU situation to me.

 

Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

http://www.sheltonjohns.com http://www.sheltonjohns.com/ 





 

On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:





I have a group of users whos complaint about Asterisk is that the
directory

application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to create
a

custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more verbose?
We

go by first name. 

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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Mark Coccimiglio]

Mark,

Do these 1600 series Cisco routers you mention that you find on eBay 
for $50-$150 support Layer3 routing?  I have a managed switch setup on 
my home network with several VLANs defined. (work subnet, home subnet, 
VOIP subnet)   I currently have to use a Linux box to route between 
the VLANs.  I'd like to move to Gigabit routing, but I'd need to 
replace the Linux box(more processor power and new NICs) and that gets 
expensive.


I'd much rather have a router or smart switch for that matter that 
does Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed



Ed,
  Layer3 routing is a fundamental function of a router which is 
supported by the Cisco 1600 series (1605R specifically) router.  However 
VLAN definitations are not supported in the 1600 series.  You would need 
to moveup to the 1700 or 2500 series for that function.  As for Gigabit 
support the 1600 and 1700 series do not support that high speed 
interface.  These router are designed around WAN style routing operating 
at ~1.5Mbps.  Gigabit routing is a rather cutting edge capablity that is 
only seen in newer hardware.  I would checkout a Cisco Catalyst 3500 
series for something like that.  Be carefull and look closely some 
systems only support 2 ports on 1000baseT and the rest are 100BaseT.


Good luck and happy hunting,

Mark Coccimiglio


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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.

 
James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Wednesday, January 10, 2007 10:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

On Wed, Jan 10, 2007 at 09:39:47AM -0700, Administrator wrote:
 Results From cat /proc/zaptel/*
 
 Span 1: WCTDM/0 Wildcard TDM2400P Board 1
 IRQ misses: 24
 
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS (In use)
3 WCTDM/0/2 FXOKS (In use)
4 WCTDM/0/3 FXOKS (In use)
5 WCTDM/0/4 FXOKS (In use)
6 WCTDM/0/5 FXOKS (In use)
7 WCTDM/0/6 FXOKS (In use)
8 WCTDM/0/7 FXOKS (In use)
9 WCTDM/0/8 FXOKS (In use)
   10 WCTDM/0/9 FXOKS (In use)
   11 WCTDM/0/10 FXOKS (In use)
   12 WCTDM/0/11 FXOKS (In use)
   13 WCTDM/0/12 FXOKS (In use)
   14 WCTDM/0/13 FXOKS (In use)
   15 WCTDM/0/14 FXOKS (In use)
   16 WCTDM/0/15 FXOKS (In use)
   17 WCTDM/0/16 FXOKS (In use)
   18 WCTDM/0/17 FXOKS (In use)
   19 WCTDM/0/18 FXOKS (In use)
   20 WCTDM/0/19 FXOKS (In use)
   21 WCTDM/0/20 FXOKS (In use)
   22 WCTDM/0/21 FXOKS (In use)
   23 WCTDM/0/22 FXOKS (In use)
   24 WCTDM/0/23 FXOKS (In use)
 Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 B8ZS/ESF
 
   25 WCT1/0/1 Clear (In use)
   26 WCT1/0/2 Clear (In use)
   27 WCT1/0/3 Clear (In use)
   28 WCT1/0/4 Clear (In use)
   29 WCT1/0/5 Clear (In use)
   30 WCT1/0/6 Clear (In use)
   31 WCT1/0/7 Clear (In use)
   32 WCT1/0/8 Clear (In use)
   33 WCT1/0/9 Clear (In use)
   34 WCT1/0/10 Clear (In use)
   35 WCT1/0/11 Clear (In use)
   36 WCT1/0/12 Clear (In use)
   37 WCT1/0/13 Clear (In use)
   38 WCT1/0/14 Clear (In use)
   39 WCT1/0/15 Clear (In use)
   40 WCT1/0/16
   41 WCT1/0/17 Clear (In use)
   42 WCT1/0/18 Clear (In use)
   43 WCT1/0/19 Clear (In use)
   44 WCT1/0/20 Clear (In use)
   45 WCT1/0/21 Clear (In use)
   46 WCT1/0/22 Clear (In use)
   47 WCT1/0/23 Clear (In use)
   48 WCT1/0/24 HDLCFCS (In use)

Is it supposed to be T1 or E1?

The card behaves as E1 but you attempt to configure it as T1.

 
 zaptel.conf file:
 
 fxoks=1-24
 span=2,1,0,esf,b8zs
 bchan=25-39
 #bchan=40
 bchan=41-47
 dchan=48
 loadzone=us
 defaultzone=us
 
 James
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
 Sent: Tuesday, January 09, 2007 11:25 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Problem with zaptel drivers or card
 
 On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
  I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
 and
  Zaptel 1.4
  
  The Digium cards installed are TDM2400 and TE110P.
  
  Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
  
  Now when I run ztcfg I get the following error message:
  
  (CAS signalling on span 2 conflicts with Clear channel on channel 40)
  
  --NOTE: signaling was spelled wrong in the error message--
  
  I have since upgraded to 1.4 with the same problem.
  
  Channel 40 is a standard bchan configuration and our provider sees no
  problem with the channel.
  
  When I disable the channel everything works fine.
  
  My assumption is that something is wrong with the TE110P card.
  
  Has anyone seen anything else like this?
 
 What do you get from:
 
 cat /proc/zaptel/*
 
 What do you have on /etc/zaptel.conf  ?
 
 -- 
Tzafrir Cohen   
 icq#16849755jabber:[EMAIL PROTECTED]
 +972-50-7952406   mailto:[EMAIL PROTECTED]   
 http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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-- 
   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] Send email notification

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 01:41:39PM -0400, H Aranguren wrote:
 Thanks for your answer Anselm,
 
   But, why do you think that the problem is in the mail server, if I
 can send mails with esmtp, with the command /usr/sbin/sendmail without
 problem. But the Voicemail app never sends the notification.

But you use a different command. Why do you need to override the default
sendmail command, BTW?

 
 I used ethereal and i couldn't see any message from asterisk box to my
 smtp server when i leave a voicemail

What is /usr/sbin/sendmail? sendmail? postfix? any other MTA?

What do you see in its logs?

-- 
   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] Send email notification

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 13:41 -0400 schrieb H Aranguren:
 Thanks for your answer Anselm,
 
But, why do you think that the problem is in the mail server, if I
 can send mails with esmtp, with the command /usr/sbin/sendmail without
 problem. But the Voicemail app never sends the notification.
 
 I used ethereal and i couldn't see any message from asterisk box to my
 smtp server when i leave a voicemail

My experience with people setting up a mail server is that they tend to
forget small but important things - I often enough do myself, with all
that complexity. Like the possibility that the sendmail command, run
as user asterisk, will not be allowed to send mail from any
e-mail-adress but [EMAIL PROTECTED] or so. If you use the sendmail -t
command as you wrote, then the first step of any e-mail to be sent will
be local and not appear in ethereal. Have you looked in the log files?
Are you _sure_ notifications are not sent?
When you replace sendmail -t with something like cat  /tmp/1, will
that file appear? Might you have a PATH issue, like sendmail living
in /usr/bin instead of /usr/sbin/?

Just guessing in the dark, and naming things that would most probably
happen in a debug session if I sat in front of the machine.

Hth
Anselm

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Yes you are correct. I do NOT plan to use it again. I have downloaded
the latest version and plan to do an install. I was hoping there might
be an rpm for it but does not seem to be. Thanks all.
Bob Rawlinson

On Wed, 2007-01-10 at 18:25 +0100, Anton Frolov wrote:
 he is probably tried to install one of these All in one Asterisk CDs, that
 effectively formats the hard drive and installs everything from scratch,
 including the OS ;)
 
 And, yes, it will happen again, if he re-runs this CD...
 
 AF.
 
 
 Doug Crompton wrote:
  Formated your hardisk... wow that is nasty, but I also cannot understand
  how that could ever happen. There must be some other HW problem going on
  or you got a hold of some really bad code.
  
  What code (source or binary) and what were you doing when that happenned?
  
  Doug
  
  On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:
  
  Thanks for the help. I was concerned because I tried once before and it
  formatted my hard disk. I wanted to be sure that did not happen again.\
  Bob Rawlinson
 
  On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
  Has anyone heard of a build or instructions for installing Asterisk on a
  Suse 10.1 system?
  Bob Rawlinson
 
 
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  k
  
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Re: [asterisk-users] dundi ENCREJ

[Alex Robar]

Hi Ramon,

Please post your peer details from dundi.conf so we can see what your setup
is.

Also, have you tried regenerating your keys? I wound up generating my keys
twice, they just didn't work the first time, I'm not sure why.

Alex

On 1/10/07, Ramon Schönborn [EMAIL PROTECTED] wrote:


hi list,

i have the same problem as mentioned here:

http://forums.digium.com/viewtopic.php?t=2678view=nextsid=bd94cefd823b23156c5748843febb3ab

my asterisk version is 1.2.12.1

any ideas?






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--
Alex Robar
[EMAIL PROTECTED]
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[asterisk-users] Zap calls

[Jay Moore]

I have 8 Zap channels, 25-32, all of which have their own line.

My zapata.conf file looks similar to:

group=1
context=context_1
signalling=fxs_ks
channel = 25

group=2
context=context_2
signalling=fxs_ks
channel = 26

and so forth for all 8 lines, where each channel has its own group and 
incoming context.


The first 4 channels are our primary trunk lines.  If we have to make an 
 outgoing call on a trunk line, how can I have it pick the first 
available line of the 4?


My first thought would be to have another group that includes the first 
4 channels, and then use that group in the Dial() command like so:


group=9
context=whatever
signally=fxs_ks
channel = 25-28

and
Dial(Zap/g9/${EXTEN},60)


Can I repeat channels like that or will it cause Asterisk to choke?  If 
I can't do it that way, can someone suggest a way to do it?


Thanks in advance,
Jay
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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Jon Pounder]

Quoting Mark Coccimiglio [EMAIL PROTECTED]:




Mark,

Do these 1600 series Cisco routers you mention that you find on eBay 
for $50-$150 support Layer3 routing?  I have a managed switch setup 
on my home network with several VLANs defined. (work subnet, home 
subnet, VOIP subnet)   I currently have to use a Linux box to route 
between the VLANs.  I'd like to move to Gigabit routing, but I'd 
need to replace the Linux box(more processor power and new NICs) and 
that gets expensive.


I'd much rather have a router or smart switch for that matter that 
does Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?


Do you actually know how utilized the linux box is now ? its probably 
near zero

and all you need is a couple cards. Routing even with complex rules takes very
little cpu.







Thanks,
Ed



Ed,
  Layer3 routing is a fundamental function of a router which is 
supported by the Cisco 1600 series (1605R specifically) router.  
However VLAN definitations are not supported in the 1600 series.  You 
would need to moveup to the 1700 or 2500 series for that function.  
As for Gigabit support the 1600 and 1700 series do not support that 
high speed interface.  These router are designed around WAN style 
routing operating at ~1.5Mbps.  Gigabit routing is a rather cutting 
edge capablity that is only seen in newer hardware.  I would checkout 
a Cisco Catalyst 3500 series for something like that.  Be carefull 
and look closely some systems only support 2 ports on 1000baseT and 
the rest are 100BaseT.


Good luck and happy hunting,

Mark Coccimiglio


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Jon Pounder

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  _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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


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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Hi,

I'm trying to write a AGI in PHP to get the numbers dialed (with 
read()), save it into a variable to insert it into a SQL server 
database. But I cannot see results into the variable, it always return 
NULL.

Here is a piece of the AGI.

fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));

Even if I only show my_var value or try to use it inside asterisk, the 
value is NULL.

There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.



I'm not a php guy, but aren't we missing the part that retrieves the 
value saved into my_var from the call to READ?


// In this part you run the read command and asterisk
// stores the value into the channel variable my_var

fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);

// In this part you are constructing your sql statement
// with a null value cause you didn't make a call to
// GET VARIABLE before constructing your sql.

$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));

--

Warm Regards,

Lee

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Re: [asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

On 1/10/07, Yuan LIU [EMAIL PROTECTED] wrote:


Then there must be an error somewhere.  The variable READ() in Asterisk
should be usable.  Should be able to use SayDigits() to play it back - or
no
value is read.

Yuan Liu



Hi Yuan and Anton,

Let's put here all AGI for test:

#!/usr/bin/php -q
?php
ob_implicit_flush(false);
error_reporting(0);
$stdin = fopen( 'php://stdin', 'r' );

if (!defined('STDIN'))
{
   define('STDIN',fopen('php://stdin','r'));
}
if (!defined('STDOUT'))
{
   define('STDOUT',fopen('php://stdout','r'));
}
if (!defined('STDERR'))
{
   define('STERR',fopen('php://stderr','r'));
}

while(!feof($stdin))
{
   $temp=trim(fgets(STDIN,4096));
   if (($temp==) || ($temp=\n))
   {
   break;
   }
   $s=split(:,$temp);
   $nome=str_subst(agi_,,$s[0]);
   $agi[$nome]=trim($s[1]);
}

foreach($agi as $chave=$valor)
{
   fwrite(STDERR,--$chave=$valor\n);
   fflush(STDERR);
}
$my_var=123;
fflush(STDERR);
fwrite(STDERR,Just testing\\\n);
fflush(STDERR);
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,exec saydigits ${my_var} \n);
fflush(STDOUT);

$conn=odbc_connect('MSSQL', 'asterisk', '123456');
$query = odbc_exec($conn, INSERT INTO usuario(nome) VALUES('$my_var'));
?

If I not startup $my_var=123; Saydigits receives a NULL as options. And so
nothing was inserted into db.
I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed through
it directly like Joel Lansden Joel AT digitalparadise DOT net reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] Problem with zaptel drivers or card

[Joel Vandal]

Administrator a écrit :

It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
  
I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.


--
Joel Vandal, CTO
ScopServ Inc.

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
 Yes you are correct. I do NOT plan to use it again. I have downloaded
 the latest version and plan to do an install. I was hoping there might
 be an rpm for it but does not seem to be. Thanks all.
 Bob Rawlinson

suse 10.1 actually includes a package of Asterisk 1.2.5 . 10.2 includes
1.2.13 . I have no idea if security updates bothered updating 1.2.5 .

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

[Carlos Chavez]

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped.  Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.

There are 9 lines connected to 3 TDM04B cards.  The Panasonic Pbx we
replaced did not have this problem.  There are 8 SIP phones and 16
analog phones connected to two Astribank-8 units and everyone claims
that their calls are dropped several times a day.

Any suggestions?  Here is my zapata.conf:

language=es
context=default
;rwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=yes
;echocancelwhenbridged=yes
rxgain=-1.0
txgain=0.0
busydetect=yes
callprogress=no
accountcode=Telmex
amaflags=default
signalling=fxs_ls
group=1
faxdetect=none
callerid=asreceived
channel = 1-6

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001


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[asterisk-users] SIP invite and sip.conf relationship?

[Tony Mountifield]

I'm having a bit of trouble setting up my sip.conf entries to accept
calls from a particular provider, and the problem really is that I am
unclear exactly what parts of the INVITE are supposed to match what
parts of sip.conf.

I couldn't find this info on the wiki, so if someone here can shed
some light, I would be very grateful!

Here are the relevant lines from the INVITE (from sip debug):

-- SIP read from 213.166.5.130:5060: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
From: 01234567890 sip:[EMAIL PROTECTED];tag=2F6B6198-D3D
To: sip:[EMAIL PROTECTED]
Date: Wed, 10 Jan 2007 18:18:22 gmt
CSeq: 101 INVITE
Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off
Contact: sip:[EMAIL PROTECTED]:5060

How do the items above, such as source address, INVITE URL, From, To, etc.,
relate to items in sip.conf in a type=user section, such as [sectionname],
user=username, host=hostname or host=dynamic, etc?

My provider gives me the option to set the invite URL, such as
sip:sip.mydomain.com or sip:[EMAIL PROTECTED], but does not use
a secret to authenticate. Does the myuser part get used at all?

Thanks for any insight.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

There is certainly an rpm. Not sure about 1.4, but at least for 1.2.

AF.



Robert A. Rawlinson wrote:
 Yes you are correct. I do NOT plan to use it again. I have downloaded
 the latest version and plan to do an install. I was hoping there might
 be an rpm for it but does not seem to be. Thanks all.
 Bob Rawlinson
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Re: [asterisk-users] Get dialed numbers in AGI

[Anton Frolov]

hi

I never really programmed in PHP, I use Perl for my purposes.
I found a good AGI library for Perl and is happy with it. It allows me to call
functions instead of parsing the input.
While looking for my library, I saw at least one for PHP. So why not to use it?

In Perl it looks like:
  my %agiArgs = $AGI-ReadParse();
  my $callerNum = $agiArgs{callerid}; // Got the caller id
  $retval = $AGI-exec('Dial', $CHANNEL.|.$CALL_OPTIONS); // Placing a call

it's so simple...
(and you have the error checking built in!)

I'm sure you could find such a library for PHP as well!

AF.


Ralph Liebessohn wrote:
 On 1/10/07, *Yuan LIU* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote:
 
 Then there must be an error somewhere.  The variable READ() in Asterisk
 should be usable.  Should be able to use SayDigits() to play it back
 - or no
 value is read.
 
 Yuan Liu
 
 
  Hi Yuan and Anton,
 
 Let's put here all AGI for test:
 
 #!/usr/bin/php -q
 ?php
 ob_implicit_flush(false);
 error_reporting(0);
 $stdin = fopen( 'php://stdin', 'r' );
 
 if (!defined('STDIN'))
 {
 define('STDIN',fopen('php://stdin','r'));
 }
 if (!defined('STDOUT'))
 {
 define('STDOUT',fopen('php://stdout','r'));
 }
 if (!defined('STDERR'))
 {
 define('STERR',fopen('php://stderr','r'));
 }
 
 while(!feof($stdin))
 {
 $temp=trim(fgets(STDIN,4096));
 if (($temp==) || ($temp=\n))
 {
 break;
 }
 $s=split(:,$temp);
 $nome=str_subst(agi_,,$s[0]);
 $agi[$nome]=trim($s[1]);
 }
 
 foreach($agi as $chave=$valor)
 {
 fwrite(STDERR,--$chave=$valor\n);
 fflush(STDERR);
 }
 $my_var=123;
 fflush(STDERR);
 fwrite(STDERR,Just testing\\\n);
 fflush(STDERR);
 fwrite(STDOUT,exec read
 my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
 \n);
 fwrite(STDOUT,exec saydigits ${my_var} \n);
 fflush(STDOUT);
 
 $conn=odbc_connect('MSSQL', 'asterisk', '123456');
 $query = odbc_exec($conn, INSERT INTO usuario(nome) VALUES('$my_var'));
 ?
 
 If I not startup $my_var=123; Saydigits receives a NULL as options.
 And so nothing was inserted into db.
 I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed
 through it directly like Joel Lansden Joel AT digitalparadise DOT net
 reported on 9/14/06.
 Is there another function or way to test it or I must try in another
 asterisk box?
 
 -- 
 Ralph Liebessohn
 ICQ: 74835911
 Skype: liebessohn
 
 
 
 
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Re: [asterisk-users] Proper use of the Local channel

[Eric \ManxPower\ Wieling]

[EMAIL PROTECTED] wrote:

Is there any documentation you guys can point us to in order to learn more 
about the proper use of the Local channel? We don't currently use it. However, 
while evaluating other people's billing and management systems for Asterisk, we 
noticed they make extensive use of it.


Did you read localchannel.txt in the asterisk docs directory in the 
source tree?

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Re: [asterisk-users] RTP directly

[Eric \ManxPower\ Wieling]

David Alcott wrote:


Is there a way to configure the Asterisk so that the RTP goes directly 
between the Endpoints as opposed to going through the asterisk?


That is the default if Asterisk believes it will work.  Things that 
might not make it work is tTwW options to Dial, protocol transation (one 
leg is SIP, the other is IAX2, transcoding, NAT, or many other things 
that make the two legs of the call not compatible with reinvites.

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Re: [asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

On 1/10/07, Lee Jenkins [EMAIL PROTECTED] wrote:


Ralph Liebessohn wrote:
 Hi,

 I'm trying to write a AGI in PHP to get the numbers dialed (with
 read()), save it into a variable to insert it into a SQL server
 database. But I cannot see results into the variable, it always return
 NULL.
 Here is a piece of the AGI.

 fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);
 fflush(STDOUT);
 $conn=odbc_connect('MSSQL', 'USER', 'PASS');
 $query = odbc_exec($conn, INSERT INTO dialed(number)
VALUES('$my_var'));

 Even if I only show my_var value or try to use it inside asterisk, the
 value is NULL.
 There is another way to do it? Am I doing a mistake here?
 I'm using Asterisk 1.2.13.


I'm not a php guy, but aren't we missing the part that retrieves the
value saved into my_var from the call to READ?

// In this part you run the read command and asterisk
// stores the value into the channel variable my_var

fwrite(STDOUT,exec Read my_var|/sound_to_play|5|||15 \n);

// In this part you are constructing your sql statement
// with a null value cause you didn't make a call to
// GET VARIABLE before constructing your sql.

$query = odbc_exec($conn, INSERT INTO dialed(number) VALUES('$my_var'));

--

Warm Regards,

Lee




Hi Lee,

thanks for the tip. I tried other methods trying to get the variable value,
but no success.
Doing a GET VARIABLE my_var after READ the get variable returns the value
I dialed, but doesn't give the exact value to it. I got Resource ID #1
instead.
Using:
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);

I got it:

AGI Rx  exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
   -- AGI Script Executing Application: (read) Options:
(my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15)
   -- Accepting a maximum of 5 digits.
   -- Playing '//usr/share/asterisk/sounds/please-wait-connect-oncall-eng'
(language 'en')
   -- User entered '85214'
AGI Tx  200 result=0
AGI Rx  get variable my_var
AGI Tx  200 result=1 (85214)
AGI Rx  exec saydigits Resource id #1
   -- AGI Script Executing Application: (saydigits) Options: (Resource)
AGI Tx  200 result=0
AGI Rx  exec Resource id #1
   -- AGI Script Executing Application: (Resource) Options: (id)
Jan 10 17:31:33 WARNING[4867]: res_agi.c:1147 handle_exec: Could not find
application (Resource)
AGI Tx  200 result=-2


I also tried:
$my_var=fwrite(STDOUT,get variable my_var \n);

But always I get 21 as value.
More tries?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Strange! I had checked on both my DVD and on the Suse site and I have
not been able to find it. Do you happen to know where it is located?
Bob Rawlinson

On Wed, 2007-01-10 at 21:03 +0200, Tzafrir Cohen wrote:
 On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
  Yes you are correct. I do NOT plan to use it again. I have downloaded
  the latest version and plan to do an install. I was hoping there might
  be an rpm for it but does not seem to be. Thanks all.
  Bob Rawlinson
 
 suse 10.1 actually includes a package of Asterisk 1.2.5 . 10.2 includes
 1.2.13 . I have no idea if security updates bothered updating 1.2.5 .
 

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Re: [asterisk-users] Random dropped calls...

[Tzafrir Cohen]

Hi!

On Wed, Jan 10, 2007 at 01:14:59PM -0600, Carlos Chavez wrote:
   I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
 having calls dropped.  Sometimes you can stay on the phone for a long
 time and sometimes the call is dropped within a minute.
 
   There are 9 lines connected to 3 TDM04B cards.  The Panasonic Pbx we
 replaced did not have this problem.  There are 8 SIP phones and 16
 analog phones connected to two Astribank-8 units and everyone claims
 that their calls are dropped several times a day.
 
   Any suggestions?  Here is my zapata.conf:
 
 language=es
 context=default
 ;rwink=300
 usecallerid=yes
 hidecallerid=no
 callwaiting=no
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echotraining=yes
 ;echocancelwhenbridged=yes
 rxgain=-1.0
 txgain=0.0
 busydetect=yes
 callprogress=no
 accountcode=Telmex
 amaflags=default
 signalling=fxs_ls
 group=1
 faxdetect=none
 callerid=asreceived
 channel = 1-6

Those are 6 channels of the 9?

What is the configuration of the other three?

What is the configuration of the 16 Astribank channels?

You don't set busycount. This uses the default value (3?). Can you try
setting it to a higher value?

-- 
   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] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Could you point me to where it is located? I had tried Suse and
sourceforge.
Bob Rawlinson

On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
 There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
 
 AF.
 
 
 
 Robert A. Rawlinson wrote:
  Yes you are correct. I do NOT plan to use it again. I have downloaded
  the latest version and plan to do an install. I was hoping there might
  be an rpm for it but does not seem to be. Thanks all.
  Bob Rawlinson
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Re: [asterisk-users] Asterisk and Avaya IP Office

[housi mueller]

In this case I would need to purchase an E1 card for the Avay PBX an an other 
for the *.  To save costs, I would like to intent the interconnection over the 
FXO port.
   
  Anyone has done this configuration so far?
  

Robert Boardman [EMAIL PROTECTED] wrote:
  Just done this for a client using an E1 Pri card in the avaya box and a 
sangoma a102, using qsig , works fine, I wouls recommend this to any 
oneits been up and stable for two months now

Regards
Robb

housi mueller wrote:
 The main goal is that any extension from the Avaya PBX can make long 
 distance calls using the asterisk server as VoIP gateway (using a SIP 
 Provider).
 It would be also great if from a remote IP Phone (in an other 
 location), a user could use the Asterisk server to dial in and the * 
 forward’s the call to an Avaya extension.
 The Avaya has an VCM card an IP Phones (5610) as extensions. First I 
 thought to connect the * to the Avaya through the ethernet interface 
 but then I was reading in forums that there are for Avaya third party 
 IP phone licence needed and that the communication with oh323 is not 
 stable.
 I thought also putting the Asterisk in front of the Avaya.
 Telco T1 - Asterisk - T1 - Avaya PBX
 This could be a solution for later one. Right know for testing it 
 would be to expensive. That's why I thought about the Avaya analog 
 Asterisk FXO interconnection.
 Any suggestions..?

 */Thomas Kenyon /* wrote:

 housi mueller wrote:
  I would like to connect an Asterik server to an Avaya IP Office
 IP406
  and use the * as an VoIP Gateway.
 
  The IP Office has two Analog extensions available. I thought
 connecting
  this analog extensions to 2 FXO ports in the * to interconnect
 the PBX’s.
 
 What sort of interaction are you after? It may be a better idea to
 try
 to intercept the line card with asterisk, or if the IP406 has a
 VCM card
 then to talk to it through the ethernet interface.

  Is this possible? Does any one have experience with such a
 configuration?
 
  Thanks in advance for all recommandations and suggestions..
 
  Housi Mueller

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[asterisk-users] Possibility to catch DTMF when 2 users are in a conversation

[Antoine Fressancourt]

Hello,

I will expose my problem here. Please tell me if it is not the right  
place as I am really new to that list.


I use Asterisk as a SIP proxy. I have two users connected to it,  
Let's call them 1234 and 5678


In my dialplan I have two lines:

exten = 1234,1,Dial(SIP/1234)
exten = 5678,1,Dial(SIP/5678)

The SIP phones (X-lite) are configured to send DTMF's using RFC 2833  
mechanism.


I want to know if it is possible in Asterisk to catch a DTMF event  
sent by one of the phone to trigger an action, for example to play a  
sound/video clip to one of the phones.


Thank you very much in advance for your help,

Antoine
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Re: [asterisk-users] Directory too difficult?

[Bryan M. Johns]

Exactly.

ESU = Equipment Superior to Users

;-)

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:


More like a ID-10-T error…..







From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Bryan M. Johns

Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?



I wish had some pearl of wisdom here, but I don't.  I will simply  
share my sympathy.




Sounds like an ESU situation to me.



Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

http://www.sheltonjohns.com






On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:




I have a group of users whos complaint about Asterisk is that the  
directory


application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to  
create a


custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more  
verbose? We


go by first name.

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RE: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[shadowym]

Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements. 

-Original Message-
From: Ed Rubright - mail lists [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, January 10, 2007 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Best inexpensive home office router
forVoIP (QoS with maybe PoE)

Mark Coccimiglio wrote:
 Marty,
Where are you paying $1000 for a 1600 series Cisco?  I can get you 
 20% off that price on any quantity (note: Sarcasam).  Its not the 
 1990's anymore.  You can get them on eBay ($50-150) for only slightly 
 more then the Linksys.  The performance is rock solid.  Three-quarters 
 of the world have used them for decades.  I know of units running 2 
 and 3 YEARS between reboots.  The power company reboots my equipment 
 more then I do.  Ok it is true that Cisco does not support the models 
 anymore, but you can't buy a services contract for a linksys router 
 either.  It can sometimes be a little difficult to configure without 
 any technical knowledge but that is what most of us get paid for.  It 
 does impress the customer when you bring in the grey box labled 
 Cisco.  As for performance just try to put 50 people behind a 
 linksys/netgear/dlink.  I've used 1605R supporting +100 users.  Not 
 even a blink.  Finally, untill everyone is using 10Mps FTTH the 
 broad band link is still the slowest part of the connection.  Not to 
 shabby for antiquated technology.

 Mark C

 Martin Joseph wrote:

 On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:

 Mike
 I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router 
 with Fair-Weight queueing enabled.  Works great.  The nice thing 
 about Fair-Weight queueing is that it dynamically adapts to lower 
 the priority of higher demand traffic (e.g. large downloads).  If 
 you want quality stick with quality stuff.

 Mark C


 Reread the subject line please.  $1000 (US) isn't inexpensive by any 
 stretch.

 Marty


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Mark,

Do these 1600 series Cisco routers you mention that you find on eBay for
$50-$150 support Layer3 routing?  I have a managed switch setup on my home
network with several VLANs defined. (work subnet, home subnet, VOIP 
subnet)   I currently have to use a Linux box to route between the 
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the
Linux box(more processor power and new NICs) and that gets expensive.

I'd much rather have a router or smart switch for that matter that does
Gigabit Layer3 routing all in one unit. 

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed


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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 02:48:41PM -0500, Robert A. Rawlinson wrote:
 Strange! I had checked on both my DVD and on the Suse site and I have
 not been able to find it. Do you happen to know where it is located?
 Bob Rawlinson

I simply checked the list of source RPMs availble from the first suse
mirror I could find at opensuse.org .

-- 
   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] Get dialed numbers in AGI

[Yuan LIU]

From: Ralph Liebessohn [EMAIL PROTECTED]

Hi Yuan and Anton,

Let's put here all AGI for test:

#!/usr/bin/php -q
?php

...

$my_var=123;
fflush(STDERR);
fwrite(STDERR,Just testing\\\n);
fflush(STDERR);
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,exec saydigits ${my_var} \n);
fflush(STDOUT);

$conn=odbc_connect('MSSQL', 'asterisk', '123456');
$query = odbc_exec($conn, INSERT INTO usuario(nome) VALUES('$my_var'));
?

If I not startup $my_var=123; Saydigits receives a NULL as options. And 
so

nothing was inserted into db.


I did a quick test and it seems that everything passed to AGI is by value, 
and there is no apparent relationship between variable named used in two 
different AGI commands.


However, a small adaption of dial plan could accomplish what you wanted, 
that is, to read the variable in dial plan, then pass its value to AGI.  
Hope this helps.


Yuan Liu

I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed 
through

it directly like Joel Lansden Joel AT digitalparadise DOT net reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn




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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

well, I'm not rpm user anymore for several years already... Isn't it
http://www.rpmfind.com/ that is used to find the rpms?

AF.


Robert A. Rawlinson wrote:
 Could you point me to where it is located? I had tried Suse and
 sourceforge.
 Bob Rawlinson
 
 On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
 There is certainly an rpm. Not sure about 1.4, but at least for 1.2.

 AF.



 Robert A. Rawlinson wrote:
 Yes you are correct. I do NOT plan to use it again. I have downloaded
 the latest version and plan to do an install. I was hoping there might
 be an rpm for it but does not seem to be. Thanks all.
 Bob Rawlinson
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

here it is, mainly for suse

http://rpmseek.com/rpm-pl/asterisk.html?hl=comcs=asterisk:PN:0:0:0:0

it's only one of the rpms (the basic one). You should make the search yourself
(try asterisk) to locate all of them.

AF.


Robert A. Rawlinson wrote:
 Could you point me to where it is located? I had tried Suse and
 sourceforge.
 Bob Rawlinson
 
 On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
 There is certainly an rpm. Not sure about 1.4, but at least for 1.2.

 AF.



 Robert A. Rawlinson wrote:
 Yes you are correct. I do NOT plan to use it again. I have downloaded
 the latest version and plan to do an install. I was hoping there might
 be an rpm for it but does not seem to be. Thanks all.
 Bob Rawlinson
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
 
 well, I'm not rpm user anymore for several years already... Isn't it
 http://www.rpmfind.com/ that is used to find the rpms?

It's meant to find rpm pckages not from your distribution that are not
supported and may be incompatible with it. Yeah.

-- 
   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] Proper use of the Local channel

[lists]

No, I haven't. I'll start there.

Thanks

On Wed, January 10, 2007 2:38 pm, Eric \ManxPower\ Wieling [EMAIL 
PROTECTED] said:

 [EMAIL PROTECTED] wrote:
 Is there any documentation you guys can point us to in order to learn more 
 about
 the proper use of the Local channel? We don't currently use it. However, 
 while
 evaluating other people's billing and management systems for Asterisk, we 
 noticed
 they make extensive use of it.
 
 Did you read localchannel.txt in the asterisk docs directory in the
 source tree?
 

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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

How did you, or do go about reversing the patch?

 
James 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: Wednesday, January 10, 2007 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

Administrator a écrit :
 It is a T1 and I am not sure what you mean by behaves like an E1. The
 connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
 happens that the problem channel is 16 on the card. This worked fine for
 over a year before the upgrade to the zaptel drivers.
   
I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.

--
Joel Vandal, CTO
ScopServ Inc.

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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Mark Coccimiglio]

shadowym wrote:


Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements. 

 



Buzzwrong answer!  Don't answer on things you have no idea.  and 
stop providing bad information.


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[asterisk-users] generating SIP errors

[Steve Cayona]

I have a DID vendor that wants me to be able to generate specific SIP 
error messages under certain conditions and I'm completely stumped on 
how to do these:


#1 - They want to see a SIP 503 error response(service unavailable) when 
they send the call in to an active extension and and the service is not 
available, I don't have a

clue on how to simulate this.

#2 - When they send in a call to an extension that doesn't exist they 
want to see a SIP 100 TRYING message before the receive the 404 NOT 
FOUND error.  Currently

I have only been able to generate the 404 error.

Any help, clues, tips or tricks are greatly appreciated in advance.  
I've searched the web for hours begging for scraps and still have come 
up empty handed

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[asterisk-users] Round Robin Queue

[Felipe Neuwald]

Hi Folks,

I implemented an Asterisk 1.2.10 on a Debian GNU/Linux, and I have
implemented a round robin queue (and a memory round robin queue too).

Here I have one simple problem:

- agent 1 (busy)
- agent 2 (busy)
- agent 3 (free)

When someone call to my queue, the action of the queue is this:
call agent 1, then call agent 2, and then call agent 3, that is free and
finally ring. There is someway to my queue only call free agents?

Thank you,

Felipe Neuwald.
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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Jon Pounder]

Quoting Mark Coccimiglio [EMAIL PROTECTED]:




shadowym wrote:


Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements.


Buzzwrong answer!  Don't answer on things you have no idea.  and 
stop providing bad information.


you should take your own advice  - an acre is 200ft x 200ft - what idiot would
pay a consultant $7000 to tell them they need one access point in the middle.






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Jon Pounder

  _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
   _/_/_/  _/  _/ _/_/_/  _/  _/_/
  _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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


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RE: [asterisk-users] caller id not transferred to SIP device

[Yuan LIU]

From: Tobias Unsleber [EMAIL PROTECTED]

Hello,

I'm wondering why asterisk is not transferring the callerid to the sip 
device.

Scenario as follows:

sangoma --- zaptel --- asterisk --- sip --- SIP-Device

zaptel is reporting the callerid, but in the sip packages the sip-address
shows unknown as user part, as this sip debug package shows:


Have you set up a callerid for your Asterisk box? (Could be anything.)  I 
got Asterisk as caller ID before setting callerid.  Afterward (as I recall 
the sequence of events) I get caller's ID.


Yuan Liu


Executing Dial(Zap/62-1, SIP/123|25|d) in new stack
We're at 172.31.253.80 port 10460
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP

...

--
Tobias Unsleber
VoIP Consultant



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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Hi Lee,

thanks for the tip. I tried other methods trying to get the variable 
value, but no success.
Doing a GET VARIABLE my_var after READ the get variable returns the 
value I dialed, but doesn't give the exact value to it. I got Resource 
ID #1 instead.

Using:
fwrite(STDOUT,exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15 
\n);

fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);

I got it:

AGI Rx  exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
-- AGI Script Executing Application: (read) Options: 
(my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15)

-- Accepting a maximum of 5 digits.
-- Playing 
'//usr/share/asterisk/sounds/please-wait-connect-oncall-eng' (language 'en')

-- User entered '85214'
AGI Tx  200 result=0
AGI Rx  get variable my_var
AGI Tx  200 result=1 (85214)
AGI Rx  exec saydigits Resource id #1
-- AGI Script Executing Application: (saydigits) Options: (Resource)
AGI Tx  200 result=0
AGI Rx  exec Resource id #1
-- AGI Script Executing Application: (Resource) Options: (id)
Jan 10 17:31:33 WARNING[4867]: res_agi.c:1147 handle_exec: Could not 
find application (Resource)

AGI Tx  200 result=-2


I also tried:
$my_var=fwrite(STDOUT,get variable my_var \n);

But always I get 21 as value.
More tries?



Again, I'm not familiar with php, but can you try enclosing your 
variable in either single or double quotes?  Like this?


fwrite(STDOUT,exec saydigits \$my_var\ \n);

...or whatever it is that you guys used to escape literals.  I use 
pascal mostly and it's strings are encased in single quotes so it's easy ;)


I looks almost like the php interpreter is handing over the literal 
pointer to the string instead of the string reference itself.  That is 
why I suggested the quotes around the string.


As another posted suggested, you should consider using a wrapper 
class/object if you're using PHP.  They've done all the work for you 
already.  If I had to write every single little piece of code that I 
used to develop software, I'd never get anything done!


Sorry can't help you more.  Hopefully someone with real php experience 
will see your post and give you a hand.


--

Warm Regards,

Lee

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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Using:
fwrite(STDOUT,exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15 
\n);

fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);

I got it:



Also you might try concatenating the values together like this:

fwrite(STDOUT,exec saydigits  + $my_var  + \n);

Of course, that might not be the correct operator (+) to glue together 
strings, but I bet this has something to do with it.  Your version above 
puts the variable name in the string itself and probably the php engine 
ignores it (unlike asterisk which seems to replace ${VAR} symbols within 
quotes).  So try bringing the variable out of the quoted string like the 
example that I gave above.


Just another suggestion.

--

Warm Regards,

Lee

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Re: [asterisk-users] Problem with zaptel drivers or card

[Joel Vandal]

Hi,

How did you, or do go about reversing the patch?
  

I have put the patch (simple) available at :

http://www.scopserv.com/download/patches/zaptel-1.2.12-reverse7860.patch

Go on your zaptel src directory and do :

patch -p0  zaptel-1.2.12-reverse7860.patch


It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
  

I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.


  


--
Joel Vandal, CTO
ScopServ Inc.
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RE: [asterisk-users] Directory too difficult?

[Colin Anderson]

I got a requirement list just now, with my comments inline: (showing it just
for a giggle)
 
User requirement: 1) Directory set up by name - If person calling does not
know employee's name, how will they access? 
 
-Why, using app_telepathy.so of course!
 
User requirement: 2) Directory set by first /or last name?? 
 
-Yes. Now decide which one.
 
User requirement: 3) Not all mobile phones have the albphabet on their
dialpads, how do they access our directory? 
 
-Shout really loud. Telus should have a class action against it for
selling Razrs with no DTMF.
 
 
 

-Original Message-
From: Bryan M. Johns [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 10, 2007 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?


Exactly. 

ESU = Equipment Superior to Users

;-)


Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
 http://www.sheltonjohns.com/ http://www.sheltonjohns.com


On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:


More like a ID-10-T error…..








  _  


From: [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] ] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?



I wish had some pearl of wisdom here, but I don't.  I will simply share my
sympathy.



Sounds like an ESU situation to me.





Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

 http://www.sheltonjohns.com/ http://www.sheltonjohns.com







On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:





I have a group of users whos complaint about Asterisk is that the directory

application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to create a

custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more verbose? We

go by first name. 

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

but some of the packages are labeled to be for SuSe 10.1 ...

AF.


Tzafrir Cohen wrote:
 On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
 well, I'm not rpm user anymore for several years already... Isn't it
 http://www.rpmfind.com/ that is used to find the rpms?
 
 It's meant to find rpm pckages not from your distribution that are not
 supported and may be incompatible with it. Yeah.
 
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Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

What is needed is a family of astdb manipulation commands:
astdbput family key value
astdbget family [key]
astdbdel family [key]

any others?


- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.3 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQEVAwUBRaVpXUtP/KMNOfRbAQLagQf+LVq3xPgwatLShzkm53+Uy+/oRN3IfnY6
bW1OcO1fhy0uhQXVY9BysDiJxvryqCOZBNMQqGpeqQA9jzvAVuGxf7heJeqDSeo4
hfidqyW+o2N1VtvhLEKNLsxucgZ76dzkvnKv6+zPVtOArSc4XTMveDFMj6CSM5yQ
3ljCzCSZpNviZjZpSXAIo3PozKaKlWJtMw9FyBQP2BPzULIOVR2VAaq4T7jEyFoT
tT5PUIRKUIzRuCRUBR+2DPdRZeif+RGd8vb9ScOROFiMmmuIxLy4UpGjFRuJajaM
pxLO2rAgLnWVhGzXQMCk6gx1hj0ovP63hXmEpUrScCR7q2J479XwGQ==
=kECV
-END PGP SIGNATURE-

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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Mark Coccimiglio]

Jon Pounder wrote:



you should take your own advice  - an acre is 200ft x 200ft - what 
idiot would
pay a consultant $7000 to tell them they need one access point in the 
middle.




I have a BA in Electronic Engineering, a Masters in Computer Science and 
I'm an FCC licensed

radio operator.  I think I know what I'm talking about.

 Life isn't always as simple as that.  What if  its a warehouse that is 
60x800ft.  still about an acre
(I've seen this one myself).  How will the system perform once the empty 
space is occupied with
inventory?  How will metal shelving effect performance.  What hardware 
should you use?  Netgear,
dLink, Linksys, Cisco (they are different), Alvarion, Proxima? If its an 
outside area an AP in the
middle is not necessarly practicle.  You can't just use any antenna 
combination you want  There are
rules governing use.  Are you certified to assemble and test such a 
system for Part 15 compliance?
Do you know the specs and ERP limits?  Who has presidence FCC or OSHA 
regs?  What about
other ISM bands?  How long can you make your ethernet runs or should you 
use Fiber Optics? 
These are the types of things that an Engineer addresses. 

...one access point in the middle.   It may work it may not.  One 
thing for sure is that the system

probably won't perform as you expect it.

Mark C


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Re: [asterisk-users] Asterisk build for Suse 10.1

[Hans Witvliet]

On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
 Has anyone heard of a build or instructions for installing Asterisk on a
 Suse 10.1 system?
 Bob Rawlinson
 
 
Hi Bob,

Afair, asterisk was not on the cdrom's (which some people use to make
their own dvd), but it was on the original DVD, aswell as on several
ftp-sites.
As long as you don't intend to use isdn-bra (isdn-2) the rpm's seems to
contain all you need.
Not just the binaries, but also about 50 config files in /etc/asterisk
you need to configure. The graphical config-tool yast won't assist you
here... Neither for 10.2, nor will there be for 10.3 :-(
(btw, for 10.2 is asterisk-1.2 included on the cdrom's)

Hans

-- 
pgp-id: 926EBB12
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[asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[M.Hockings]

Dumpolid Exeplish wrote:

It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you

http://rfdesign.com/mag/radio_field_trials_allsoftware/



That is more what I was thinking of but it is still a cell provider type 
of hardware.  In my mind I was thinking of something very low powered 
and turning off the roaming, etc on the phone so they only work with the 
one base.   Think single cell base-station transceiver that can talk to 
a cell phone and turn it into a sip conversation to Asterisk.  Here in 
Canada, and back years ago, when I worked with radio I think the law was 
something like less than 100mw of input power didn't require a license. 
 However, with the advent of cell phones that could very well not be 
the case in those bands.  But one never knows...


In any case I'll probably lean towards something like the Engenius 
wireless phones.


http://www.engeniustech.com/telecom/products/details.aspx?id=107

But it would have been slick to be able to use the old analogue cell 
phones as we have several unused here and I am sure they would be cheap 
or free to pick up more.


Thanks to all for your input

Mike

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