date:20071031

Re: [asterisk-users] Druid

2007-10-31 Thread Alex Epshteyn

Dean,

 

If you are looking for a non-restricting and extensible Asterisk GUI please
look at Thirdlane http://www.thirdlane.com  . If
you are comfortable installing OS, Webmin and Asterisk, I would suggest
installing PBX Manager GUI (packaged as a Webmin module), otherwise
Thirdlane Advantage (CentOS based ISO) may be a good option.  

 

Best regards,

Alex 

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Wednesday, October 31, 2007 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Druid

 

Is anyone out there using Druid?

 

After the switchbox announcement today I've been looking into some other
gui's and as I'll probably do a trial install this weekend of the free
switchvox iso but I thought I'd ask is there any other guis I should be
burning trial ISO's of as well?

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
  [EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

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Re: [asterisk-users] flooded by "Maximum trunk data space exceeded" messages

2007-10-31 Thread Louis-David Mitterrand

On Wed, Oct 31, 2007 at 04:53:49PM +0400, Arun Kumar wrote:
> try to reduce number of calls on trunk or create multiple trunks.

The flood happens when I have only one call on the trunk.

> On 10/31/07, Louis-David Mitterrand <[EMAIL PROTECTED]>
> wrote:
> >
> > Hi,
> >
> > Using 1.4.13 and trunking a single iax channel to a similar box my
> > asterisk console is flooded with:
> >
> > [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data
> > space exceeded to xx.xx.xx.xx:4569
> >
> > Known issue?
> >
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Re: [asterisk-users] G.729 required for IP<--->TDM<--->IP

2007-10-31 Thread satish patel

Thanks 

 But is there any voice qulity effect if u use free version G.729 codec or 
license codec ???

If u go for license then any effect on voice qulity 


"[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: Here's a link to the "free" 
version:

http://asterisk.hosting.lv/


On 10/31/07, Gordon Henderson  wrote:
> On Tue, 30 Oct 2007, satish patel wrote:
>
> > Dear all
> >
> > I have already post this question but i need more input for 
> > this setup
> >
> > [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention]
> >
> > Asterisk - codec (G.711/ulaw)
> > Avaya - codec ( G.711/ulaw)
> >
> > Now I need G.729 on my asterisk side and i have put G.729 codec setting
> > on my IP phone and when i make call from asterisk to Avaya Extention i
> > got error
> >
> > translator not in path
> >
> > so i need to get license of g.729 on asterisk for transcoder or it will
> > work wothout translator ???
> >
> > My question is :-- Is there Required G.729 (License) on Asterisk Or Not
> > ???
>
> You can purchase them from Digium:
>
> http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5
>
> $10 each.
>
> Install one license for each simultaneous g792 call you expect to take on
> the asterisk box and off you go.
>
> There are free versions of g729 avalable, but if your country is
> compatable with the various (US) patent laws then you ought to pay the
> license fee to stay legal.
>
> Gordon
>
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PGP Signature--

Satish Patel
mobile:- +91-9818875535

http://www.linuxbug.org
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Re: [asterisk-users] hostname in MySQL CDR records

2007-10-31 Thread Steve Edwards

On Wed, 31 Oct 2007, Jim Gottlieb wrote:

> I would like to send the CDR records from all our machines around the
> world to a single database.  But I need the hostname included with each
> record for monitoring purposes.
>
> Is there a better way than using the userfield and adding
> SetCDRUserfield for every call to set the userfield to the name of the
> host?

Personally, I think the "userfield" is a hack. I prefer to add properly 
named columns to the cdrs table using cdr_addon_mysql. It makes everything 
so much more obvious -- especially when you don't have to cram several 
values into the singularly obtuse userfield.

I prefer to "retrieve" the CDRs rather than "send" them. This way, you 
only have a single script to retrieve all of the remote CDRs and the 
script is simpler since you don't have to "poll" a directory and try to 
figure out if the remote transfer has finished so you don't process a 
partial file. It also makes it easier to handle a remote host that is 
temporarily unavailable.

I retrieve the CDRs from remote hosts using a script that looks like this 
snippet:

# for each host
 for HOST in ${HOST_LIST}
 do

# mark the records to be exported
 mysql\
 ${USER_AUTH}\
 --database=mumble\
 --execute="update cdrs set disposition = 'EXPORTING'"\
 --host=${HOST}

# dump the cdrs
 mysqldump\
 ${USER_AUTH}\
 --host=${HOST}\
 --no-create-info\
 --skip-opt\
 --where="disposition = 'EXPORTING'"\
 mumble\
 cdrs\
 >/tmp/${HOST}.sql

# load the cdrs into our database
 mysql\
 ${USER_AUTH}\
 --database=mumble\
 --host=localhost\
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Re: [asterisk-users] hostname in MySQL CDR records

2007-10-31 Thread Tilghman Lesher

On Wednesday 31 October 2007 19:49:24 Jim Gottlieb wrote:
> I would like to send the CDR records from all our machines around the
> world to a single database.  But I need the hostname included with each
> record for monitoring purposes.
>
> Is there a better way than using the userfield and adding
> SetCDRUserfield for every call to set the userfield to the name of the
> host?

If you set systemname in asterisk.conf, that prefix will become part of the
uniqueid field.  You'll probably need to widen that field, though.

-- 
Tilghman

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[asterisk-users] Connection astrisk to a RAS (portmaster)

2007-10-31 Thread Nicolas Ross

Here's my planed setup :

PRI from telco <--> (port 1 of A104d) * (port 2 of A104d) <--> PM3

The PM3, for those who don't know is lucent's portmaster RAS dial-up router.

I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards).

In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2 
is TDM_VOICE, without hw echo, Clock as master, reference 0 (for the time 
being, I'm still not hooked up with my pri, will be 1 in the future), span 
2. I alswo had to enable High Impedance on port 2 to operate without alarms.

Zaptel.conf:

loadzone=us
defaultzone=us
#Sangoma A104 port 1 [slot:12 bus:0 span: 1]
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#Sangoma A104 port 2 [slot:12 bus:0 span: 2]
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

zapata.conf (part of) :

;Sangoma A104 port 1 [slot:12 bus:0 span: 1]
switchtype=national
pridialplan=unknown
context=demo
group=1
signalling=pri_cpe
channel => 1-23

;Sangoma A104 port 2 [slot:12 bus:0 span: 2]
switchtype=national
pridialplan=unknown
context=demo
group=2
signalling=pri_net
channel => 25-47



Now, after start wanpipe and asterisk, my PM3 shows that the line is up (via 
a t1 cross-over cable). I see that the channels are idle and waiting.

On * console, I get :

Primary D-Channel on span 2 up

All the time

I also get sometime :

  == Primary D-Channel on span 2 down
[Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No 
D-channels available!  Using Primary channel 48 as D-channel anyway!
  == Primary D-Channel on span 2 up
[Oct 31 20:51:05] ERROR[10250]: chan_zap.c:8178 zt_pri_error: !! Got reject 
for frame 1, but we only have others!


In my extensions.conf, I have :

exten => 1234567,1,Dial(Zap/g2/${EXTEN})

Whem I trie that extension via a softphone, I hear hald a ring, and nothing 
else. The d-channel up continue to appear on the console.

Any help would be appriciated.

Nicolas 


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Re: [asterisk-users] AEL2 and Callbacks

2007-10-31 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

What do you get if you do dialplan show default?

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHKUlTDQNt8rg0Kp4RAteuAJ9kbLC77Bw7G789uOIaQ1hR+++87gCgqNPB
p4jMkvOg6kuVylFKaHLPwAs=
=ajMg
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[asterisk-users] hostname in MySQL CDR records

2007-10-31 Thread Jim Gottlieb

I would like to send the CDR records from all our machines around the
world to a single database.  But I need the hostname included with each
record for monitoring purposes.

Is there a better way than using the userfield and adding
SetCDRUserfield for every call to set the userfield to the name of the
host?

Thanks...


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[asterisk-users] iax: they hear me, but I don't hear them...iptables?

2007-10-31 Thread sean darcy

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk
server at work from home.

I've setup zoiper for iax, set the ip address to work's fixed ip
address, user: home, password: password

but the zoiper log shows:
11:02:35 Rejected registration  for '[EMAIL PROTECTED]>' with
cause 'facility rejected'
11:03:35 Rejected registration  for 'home@' with
cause 'facility rejected'

and on the asterisk server at work I get:

 NOTICE[5072]: chan_iax2.c:5252 register_verify:  No registration for
peer 'home' (from )


iax.conf:

[general]
jitterbuffer=yes
tos=ef
iaxcompat=yes

[iax-from-home]
type=user
allow=gsm
username=home
secret=password
context=long-distance
;;auth=md5,plaintext

[iax-to-home]
type=peer
allow=all
username=home
secret=password
;;auth=md5,plaintext

[guest]
type=user
context=long-distance
callerid="Guest IAX User"

This seems like a NAT problem, but how can that be with iax if they
can hear me? Doesn't everything travel on the same port? How does my
phone ring?

Iny any event, my home iptables -L:

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
ACCEPT all  --  192.168.2.0/24   anywhere
ACCEPT all  --  10.0.0.0/8   anywhere
ACCEPT udp  --  anywhere anywhereudp dpts:1024:65535
ACCEPT udp  --  anywhere anywhereudp
dpt:avt-profile-1
ACCEPT udp  --  anywhere anywhereudp dpt:sip
.

Any help appreciated

sean

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Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls

2007-10-31 Thread Barry D. Hassler

We park the calls by transferring to extension 7000, which is our parking
extension. We have both Zap and SIP extensions, and I haven't been able to
see a pattern if its related to one or the other. The primary person
answering the phone is using a SIP phone (Grandstream GXP-2000), we have a
small number of analog phones left (2), and other SIP phones (mostly
Polycom).

The only clue I've seen with the CLI is that I'll generally see a LOT of
entries for active channels on extension 7000 if I do a "show channels".
I'll try to catch this situation again and grab the output.

I only started having this problem when I upgraded to the 1.4 version from
1.2.

On 10/31/07, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]>
wrote:
>
> Barry D. Hassler wrote:
> > I've tried to find other threads with this same topic, but haven't
> > found any... Apologies if this already being discussed
> >
> > Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4.
> >
> > Having an issue with (I think) parked calls. We tend to park calls,
> > but we're often not able to pick them back up, or the other party says
> > they get dropped, etc. There doesn't seem to be a specific pattern
> > that I've discovered so far. I had this happen to me personally this
> > morning -- receptionist parked a call for me on extension 7001, but
> > when I dialed 7001, just got dead air. I could see in asterisk that
> > the call was indeed parked though, and after calling the person back,
> > he reported he was just hearing the lovely on-hold music.
> >
> > Is there a known issue (and even better, a fix) for this situation?
> > Any other information I can provide I'll do so!
> What kind of phones are you using? are they Zap or SIP?
>
> Can you provide a CLI output with any tips in it?
>
>
>
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-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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[asterisk-users] Call Failed

2007-10-31 Thread Robert La Ferla

After so many rings when the party does not answer, my SIP phone says  
Call Failed.  Why doesn't it just keep ringing?

Here's the dial plan rule:

exten => _NX,1,Dial(SIP/[EMAIL PROTECTED],,r)
exten => _NX,n,Hangup()



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[asterisk-users] Druid

2007-10-31 Thread Dean Collins

Is anyone out there using Druid?

 

After the switchbox announcement today I've been looking into some other
gui's and as I'll probably do a trial install this weekend of the free
switchvox iso but I thought I'd ask is there any other guis I should be
burning trial ISO's of as well?

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

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[asterisk-users] AEL2 and Callbacks

2007-10-31 Thread Douglas Garstang

I am originating a command via the AMI with this...

Action: Login
Username: xxx
Secret: yyy

ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120

My LegA context:
---
context LegA {
_X. => {
Dial(SIP/[EMAIL PROTECTED]); 
}

}

And my default context:
--
context default {
callback => {
NoCDR();
Wait(1);

Dial(${destination},60,oL(${timeout}:${timeout_warning}:${timeout_warning_repeat}));
}

}

The A leg is established, and once Asterisk goes to dial the B leg...

-- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2", 
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Provider-09a8cff8 is making progress passing it to Local/[EMAIL 
PROTECTED],2
-- SIP/Provider-09a8cff8 answered Local/[EMAIL PROTECTED],2
  == Starting Local/[EMAIL PROTECTED],1 at default,callback,1 failed so falling 
back to exten 's'
  == Starting Local/[EMAIL PROTECTED],1 at default,s,1 still failed so falling 
back to context 'default'
[Oct 31 01:57:07] WARNING[29795]: pbx.c:2450 __ast_pbx_run: Channel 
'Local/[EMAIL PROTECTED],1' sent into invalid extension 's' in context 
'default', but no invalid handler

Uhm, why? I have a default context with a callback extension. Of course I have 
no explicit priority 1 though... this is AEL2 
What's it complaining for?

Doug.








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Re: [asterisk-users] Mark Spencer on Pulver TV

2007-10-31 Thread Baji Panchumarti

On 10/31/07, randulo  wrote:

> May be of interest to you:
>
> http://www.blogtv.com/Shows/96/YeTrZe3vb2V&pos=ancr

 not long enough !

 Hot tip :  you can skip the first 6 minutes

 thnx !

--

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Re: [asterisk-users] T.38 Faxing and Asterisk

2007-10-31 Thread Paul Bryson

[EMAIL PROTECTED] wrote:
> I thought there was some talk of getting T38Gateway into asterisk_addons?
> 
> Stupid "linking" bullshits.

Stupid indeed.  I'm surprised T.38 support isn't a higher priority for 
Digium, given that faxing has such a high failure rate with VoIP.


Paul


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[asterisk-users] h323 help

2007-10-31 Thread Jiann-Ming Su

We've configured ooh323 on our 1.4.6 asterisk server.
We've looked at various sites for tips, most recently
http://www.tek-tips.com/viewthread.cfm?qid=1243330&page=3.  The module
seems to load properly.  When we do a tcpdump, we see traffic flowing
between the asterisk server and the Avaya communication manager.
However, we're not geting phone calls connect.  Since we do not manage
the Avaya CM, how can we further verify that our ooh323 config is
correct?  Thanks for any tips.


-- 
Jiann-Ming Su
"I have to decide between two equally frightening options.
 If I wanted to do that, I'd vote." --Duckman
"The system's broke, Hank.  The election baby has peed in
the bath water.  You got to throw 'em both out."  --Dale Gribble
"Those who vote decide nothing.
Those who count the votes decide everything."  --Joseph Stalin

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[asterisk-users] Problem with flash hook

2007-10-31 Thread Janardhanan S

Hi,

I facing a problem with flash hook. When ever I do a flash hook to place an
extsing call on hold, the call gets disconnected. The debugs on Asterisk
shows that 'on hook event detected'  when I press the flash button on the
phone. The setup is like this

Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD
and configured for ISDN PRI lines. Analog phones come out of the FXS cards.


Phone 1
Phone 2 --FXS card . Asterisk. T1 card ISDN PRI=IAD
+IP
Phone 3

The Asterisk is configured to plce calls on one of the T1 ports/channels
based on the dial plan. ie, if I dial 94XX then the call will be
placed  on a channel in the 4th T1 port which is connected to the IAD. I am
able to make calls using this setup. But I am not able to put the call on
hold using a flash hook. The call gets disconnected when the flash button is
pressed.

I have transfer=yes, threewaycalling=yes etc enabled in my zapata.conf .

signalling=fxo_ks
threewaycalling=yes
transfer=yes
callwaiting=yes
flash=1000
context=from-internal
group=1
callgroup=1
pickupgroup=1
hidecallerid=no
usercallerid=yes
musiconhold=default
channel => 97

Here are the asterisk debugs that I get when I flash the call
-

Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Exception on 106, channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Got event On hook(1) on channel 97
(index 0)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Enabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Echo cancellation already on
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Unlinking slave 73 from 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Removed 83 from conference 9/97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Removed 106 from conference 9/73
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 97, with 0
conference users
Oct 25 10:27:13 DEBUG[28965] channel.c: Returning from native bridge,
channels: Zap/97-1, Zap/73-1
Oct 25 10:27:13 DEBUG[28965] channel.c: Hanging up channel 'Zap/73-1'
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: zt_hangup(Zap/73-1)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option AUDIO MODE, value: ON(1)
on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Hangup: channel: 73 index = 0,
normal = 83, callwait = -1, thirdcall = -1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Not yet hungup...  Calling hangup
once with icause, and clearing call
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 73
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 73, with 0
conference users
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/73-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 73
Oct 25 10:27:13 VERBOSE[28965] logger.c: -- Hungup 'Zap/73-1'
Oct 25 10:27:13 DEBUG[28965] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Oct 25 10:27:13 DEBUG[28965] app_macro.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'Zap/97-1' in macro
'dialout-trunk'
Oct 25 10:27:13 DEBUG[28965] pbx.c: Spawn extension
(macro-dialout-trunk,s,14) exited non-zero on 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28974] app_queue.c: Device 'Zap/73' changed to state
'0' (Unknown) but we don't care because they're not a member of any queue.
Oct 25 10:27:13 DEBUG[28965] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Oct 25 10:27:13 DEBUG[28965] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2007-10-25
10:26:53','6782363001','6782363001','946782362001','from-internal',
'Zap/97-1','Zap/73-1','Dial','ZAP/g4/6782362001',20,17,'ANSWERED',3,'','
1193322403.4539')
Oct 25 10:27:13 DEBUG[28965] channel.c: Hanging up channel 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: zt_hangup(Zap/97-1)
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Hangup: channel: 97 index = 0,
normal = 106, callwait = -1, thirdcall = -1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: disabled echo cancellation on
channel 97
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/97-1
Oct 25 10:27:13 DEBUG[28965] chan_zap.c: Updated conferencing on 97, with 0
conference users
Oct 25 10:27:13 VERBOSE[28965] logger.c: -- Hungup 'Zap/97-1'
Oct 25 10:27:13 DEBUG[28976] app_queue.c: Device 'Zap/97' changed to state
'0' (Unknown) but we don't care because they're not a member of any queue.
Oct 25 10:27:14 DEBUG[3779] chan_zap.c: Monitor doohicky got event Hook
Transition Complete on channel 97
Oct 25 10:27:14 DEBUG[3779] chan_zap.c: Monitor doohicky got event
Ring/Answered on channel 97
Oct 25 10:27:14 WARNING[3779] chan_zap.c: zt hook fai

Re: [asterisk-users] (no subject)

2007-10-31 Thread Tim Sharp

We have Cisco 9760 for executives and Aastra 9112i for everybody else.  

We started with Grandstream, don't remember the model, cost around $80
USD but it had bad audio quality and echo problems (running asterisk
1.09).  The quality of construction felt poor, like a toy phone.  

We replaced them with the Aastra for double the cost and the quality
improved dramatically.  Audio quality was much better and echo problems
all but eliminated.  This phone also feels more solid.  There are a few
areas that are not perfect; the speaker phone is good not excellent and
we have had to replace a couple of phones because they have stopped
working.  Over all I would say not bad for the price especially if they
are for general use. 

We had to upgrade from the Aastra phones for our executives because they
needed very good audio for both handset and speaker phone.  We are using
Cisco 9760's for them and have had no problems with quality.  Plus they
have a very solid feel.

My question to the list is:  
As I need to add phones I am considering buying used Cisco 9760's.  Is
there any difference with the 9760G?  I have heard that the 9761's have
even better audio quality.  Our main requirement is audio quality, our
users do not need a lot of features on their phones.

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Wednesday, October 31, 2007 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)

What is the issue with the Grandstream?  We are getting tired of Cisco 
issues, so we have started looking at Grandstream and they seem to be 
pretty good.  The Polycom work well, but they seem to die after about a 
year or so.  We bought 20 of them about 2 years ago and 7 of them have 
died or had buttons stop working so we had to replace them.  I haven't 
had a single Cisco do that and we have probably 100 of them.

Jim Houser wrote:
> We agree with Drew and no longer use Grandstream.   We have used a few
> Polycom, (best voice quality, hardest to configure).  I have heard
good
> things about Snom but never used them.  We standardized on Aastra.
Good
> build, sound quality, and feature set.  Easy to configure or upgrade
and
> good pricing.  If you try Snom please share your thoughts.  At present
we
> are sticking with Aastra due to good results and user feedback.
> 
> Jim
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Drew
Gibson
> Sent: Wednesday, October 31, 2007 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] (no subject)
> 
> [EMAIL PROTECTED] wrote:
>> Hi all,
>>
>> We have a client that needs to setup about 80 desk phones (about 50
in 
>> one location and about another 30 in 5 different locations). Which 
>> brand/model would you recommend. We were personally thinking in 
>> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard 
>> great things about them. However, having no real experience with them

>> makes it hard in recommending one to our customer. The only
experience 
>> we've had is a very frustrating one trying to load the IP software on

>> a Cisco 7970G and so we assume that if we have to go through that for

>> all 80 phones, we'll probably commit suicide :)
>>
>> Thanks
>>   
> 
> We have used Cisco and Aastra, can't comment on Polycom or Snom.
> 
> I cannot recommend Cisco, good sound quality but that's it.
Ridiculously
> overpriced, too few usable features, incredibly awkward to manage.
> Aastra have good sound quality, reasonable price, configs are plain
text and
> not to hard to work with. We have the 9133i as our basic phone and
480i in
> the Call Centre for the soft buttons. Both can be fed from the same
config
> templates.
> We used to use Grandstream but quality and support issues have driven
us
> away.
> 
> regards,
> 
> Drew
> 
> --
> Drew Gibson
> 
> Systems Administrator
> OANDA Corporation
> www.oanda.com
> 
> 
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Re: [asterisk-users] Faxing with Asterisk & SpanDSP [Was Fax Problems with SpanDSP]

2007-10-31 Thread Tilghman Lesher

On Wednesday 31 October 2007 13:55:32 [EMAIL PROTECTED] wrote:
> Steve Underwood wrote:
>  > SpanDSP cannot be used by the standard distribution of Asterisk, as it
>  > is GPL code. However, if you are using Asterisk within the restrictions
>  > of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
>
> I was wondering how someone could modify Asterisk to be GPL compliant?

Asterisk is already GPL compliant.  The issue is that the code has exceptions
to the GPL, to allow it to be linked to certain patented codecs, for example,
without violating patent licenses or the GPL.

SpanDSP, on the other hand, is not dual licensed, so you cannot run, for
example, both G.729 and SpanDSP in the same binary (legally), since the
pure GPL license of SpanDSP "pollutes" the Asterisk binary to require the same
of the G.729 codec (which is not permissible until the patents on G.729
expire).

-- 
Tilghman

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[asterisk-users] Asterisk configuration for T1 CAS lines

2007-10-31 Thread Janardhanan S

Hi,

I am trying to use Asterisk PBX with T1 CAS. The setup that I am looking for
is as below


Analog phones == Asterisk T1 CAS === Integrated Access Device
 IP Network for VoIP.

The Asterisk has a T1 card and  I want a CAS config between Asterisk and T1
port of IAD. The Asterisk has got a FXS card to which the analog phones are
connected.

I would like to know whether T1 CAS configuration is possible with Asterisk.
If possible, any pointers to configuration would be really helpful.

Thanks and Regards,
Jana
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Re: [asterisk-users] Mobile phone codecs ...

2007-10-31 Thread Anselm Martin Hoffmeister

Am Mittwoch, den 31.10.2007, 16:47 + schrieb Gordon Henderson:
> Not strictly asterisk related, however...
>
> No GSM!
> 
> How odd is that, given that it's a GSM mobile phone...

Maybe the GSM codec is implanted to the "GSM chip" and that one does
alaw, ulaw...

> Anyway, my quest for the ultimate "one handset" solution is getting 
> closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor 
> Granite it might have half a chance of working outside the room with the 
> access point, however ...

That is one of the two points I like at the "American style" wood and
wallpaper houses (the other being that construction is cheap and easy,
in comparison). Living in a concrete house is not all the best thing as
well though. My Pirelli dual-mode phone loses WLAN link just outside my
flat door in the hall way, one concrete wall and about five meters from
the Access Point. What luck they only used drywall inside the appartment.

> Anyone tried the Plantronics Voyager 510 bluetooth headsets which 
> regsiters to both a mobile phone and their own base unit (which 
> presumably has a USB sound device)

I had a Plantronics device here that connected to a phone-line-tap base
station or to my mobile via bluetooth. I did not buy it though because
it only worked with my Sony T610 (stone-age old, about 2003), not with
my O2 xda.

I sold one plantronics 510 to a customer who uses it with his Nokia
Esomething, and really likes it. AFAIK the USB device that comes with it
is a bluetooth dongle, not a virtual audio device, but that might be
different between versions of that device, and my customer definitely
does not use it.

I noticed with my plantronics device back then that you needed to
re-pair it (whohoo, never noticed that similarity repair to re-pair) to
whatever device you want to use it with, and that sucked because it took
half a minute and some interaction with the mobile or base station.

> as in:
> 
> https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click
> 
> I'm not a fan of soft-phones, and not sure I want to have a borg implant 
> on when I'm not driving, but ...

resistance is futile :-#

Best regards,

Anselm


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Re: [asterisk-users] Faxing with Asterisk & SpanDSP [Was Fax Problems with SpanDSP]

2007-10-31 Thread Dave Fullerton

[EMAIL PROTECTED] wrote:
> Steve Underwood wrote:
>  > SpanDSP cannot be used by the standard distribution of Asterisk, as it
>  > is GPL code. However, if you are using Asterisk within the restrictions
>  > of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
>  >
> 
> I was wondering how someone could modify Asterisk to be GPL compliant?

It *is*. The issue is that since Asterisk is dual-licensed, GPL-only 
code cannot be included in the mainline asterisk source. SpanDSP is (to 
my knowledge) GPL-only and hence, cannot be included.

-Dave

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Re: [asterisk-users] (no subject)

2007-10-31 Thread [EMAIL PROTECTED]

Honestly, Its my opinion that the Aastra phones are very lacking in
the firmware department. If they could get that sorted out I wouldn't
mind using them. But for now there are too many NAT issues mostly
caused because they use an OLD version of Broadcom CallCtrl. Why they
use an ancient version is beyond me but the phones dont even have a
NAT keepalive option. They promise updates to their firmware but then
they only fix minor bugs.

Grandstream are ok. But as others have said their support is very
lacking. I've had products of theirs behave very oddly  like
operate and refuse to apply any settings no matter what and not allow
a factory reset... paperweight.

I'd personally use Polycom in the situations where there's no NAT and
the Linksys SPA-phones where you do have NAT.

On 10/29/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> We have a client that needs to setup about 80 desk phones (about 50
> in one location and about another 30 in 5 different locations). Which
> brand/model would you recommend. We were personally thinking in
> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard
> great things about them. However, having no real experience with them
> makes it hard in recommending one to our customer. The only
> experience we've had is a very frustrating one trying to load the IP
> software on a Cisco 7970G and so we assume that if we have to go
> through that for all 80 phones, we'll probably commit suicide :)
>
> Thanks
>
>
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Re: [asterisk-users] Large voicemail

2007-10-31 Thread [EMAIL PROTECTED]

Probably the best option is store the messages in IMAP and the
userdate in a database.

Honestly I dont think there is an issue with any number of mailboxes
the issue is going to be how many calls at once your system can handle
or how well your architecture scales to handle multiple machines. Can
your storage handle 5,000 mails being recorded at once? Just trying to
sort out the thousand different aspects of it all in my mind right now
I say you give it a try but expect to write your own voicemail fron
the ground up and not necessarily based on Asterisk. Then again, I
could be wrong.

On 10/25/07, Pepo <[EMAIL PROTECTED]> wrote:
> I am trying to use Asterisk as the voicemail system of the TELCO where I work.
> I wanna test with 2 mail boxes ( and later with a better machine/server I
> hope try with 7 ).
>
> How do I include in voicemail.conf the file with the mail boxes?, In a big
> system like this,is better use text files or any database?
>
> Thanks
>
> --
>
>  Linux User Registered #232544
>   Jabber : [EMAIL PROTECTED]
>Ekiga : [EMAIL PROTECTED]
>  ICQ : 337889406
>GnuPG-key : www.keyserver.net
> ---
>dum loquimur, fugerit invida
> aetas: carpe diem, quam minimum credula postero.
>
>
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[asterisk-users] Long duration calls with Asterisk out to VoIP telco

2007-10-31 Thread Wai Wu

Hi list,

My long duration calls are being timeout by my SIP VoIP provider for
failure of receiving re-INVITE within their timeout limit. Is there a
way to config Asterisk to automatically send a re-INVITE message every
10 to 15 minutes? I looked into the sip.conf file and couldn't find such
a parameter. Thnx.



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Re: [asterisk-users] G.729 required for IP<--->TDM<--->IP

2007-10-31 Thread [EMAIL PROTECTED]

Here's a link to the "free" version:

http://asterisk.hosting.lv/


On 10/31/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
> On Tue, 30 Oct 2007, satish patel wrote:
>
> > Dear all
> >
> > I have already post this question but i need more input for 
> > this setup
> >
> > [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention]
> >
> > Asterisk - codec (G.711/ulaw)
> > Avaya - codec ( G.711/ulaw)
> >
> > Now I need G.729 on my asterisk side and i have put G.729 codec setting
> > on my IP phone and when i make call from asterisk to Avaya Extention i
> > got error
> >
> > translator not in path
> >
> > so i need to get license of g.729 on asterisk for transcoder or it will
> > work wothout translator ???
> >
> > My question is :-- Is there Required G.729 (License) on Asterisk Or Not
> > ???
>
> You can purchase them from Digium:
>
> http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5
>
> $10 each.
>
> Install one license for each simultaneous g792 call you expect to take on
> the asterisk box and off you go.
>
> There are free versions of g729 avalable, but if your country is
> compatable with the various (US) patent laws then you ought to pay the
> license fee to stay legal.
>
> Gordon
>
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Re: [asterisk-users] Faxing with Asterisk & SpanDSP [Was Fax Problems with SpanDSP]

2007-10-31 Thread [EMAIL PROTECTED]

Steve Underwood wrote:
 > SpanDSP cannot be used by the standard distribution of Asterisk, as it
 > is GPL code. However, if you are using Asterisk within the restrictions
 > of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
 >

I was wondering how someone could modify Asterisk to be GPL compliant?

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Re: [asterisk-users] T.38 Faxing and Asterisk

2007-10-31 Thread [EMAIL PROTECTED]

I thought there was some talk of getting T38Gateway into asterisk_addons?

Stupid "linking" bullshits.

On 10/31/07, Paul Bryson <[EMAIL PROTECTED]> wrote:
> Nasir Iqbal wrote:
> > Hi,
> >
> >
> > Have you tried Callweaver http://www.callweaver.org
>
> I was really hoping to be able to use Trixbox to do this and it's a
> pretty complete solution by itself.  Unfortunately that requires Asterisk.
>
> It appears that there is no way to get Asterisk, or anything on the
> Asterisk box, to act as a T.38 endpoint.  This appears to be the result
> of a licensing issue with SpanDSP.
> http://www.voip-info.org/wiki/view/T.38
>
> That's a real shame as T.38 termination support is one of the last big
> pieces for us to make Asterisk a seamless solution.
>
>
> Paul Bryson
>
>
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Re: [asterisk-users] Astribank + AsteriskNOW

2007-10-31 Thread Rupert Utteridge - Digital Techniques (Asia) Limited

You will need to go to rPath www.rpath.org and download AsteriskNOW version
6.5 beta. All should work immediately.

Rupert Utteridge

Tel:+61 2 9037 4191

Message: 18
Date: Wed, 31 Oct 2007 10:54:22 -0200
From: " Guilherme Loch Waltrick G?es " <[EMAIL PROTECTED]>
Subject: [asterisk-users] Astribank + AsteriskNOW
To: asterisk-users@lists.digium.com
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in
mantis seems to be closed, but I cannot find "fxload" or "lsusb" to do some
debugging.

-- 
Guilherme Loch G?es

MSN:[EMAIL PROTECTED]
(48) 99115299
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Re: [asterisk-users] PRI over T1 calls dropping, cause 100

2007-10-31 Thread [EMAIL PROTECTED]

On 10/31/07, Michelle Dupuis <[EMAIL PROTECTED]> wrote:
> The T1 was setup as tie line, not a trunk.   The Bell guy tried setting up
> the line 2 ways:
>
> 1.  As a trunk.  This did not work because:
>   a)  When he typed in the access code for the trunk on a phone set (and
> then any numbers), the call never appeared on the Asterisk side.
>   b)  The Bell guy said that unless Asterisk was generating a dialtone, a
> trunk would not work
>   (I struggled to understand these explanations...but figured I must be
> missing something)

There is no "dialtone" on a PRI/T1. I think what he meant was you need
to change in your zaptel config "pri_cpe" to be "pri_net" then it will
allow you to setup that trunk.

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[asterisk-users] Mark Spencer on Pulver TV

2007-10-31 Thread randulo

May be of interest to you:

http://www.blogtv.com/Shows/96/YeTrZe3vb2V&pos=ancr

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Re: [asterisk-users] Mobile phone codecs ...

2007-10-31 Thread [EMAIL PROTECTED]

On 10/31/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> Not strictly asterisk related, however...
>
> Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client
> which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?)
>
> Anyway, in a fit of idleness, I thought I'd see what codecs it supports,
> as I couldn't find it in the manual...
>
> And it supports:
>
>ilbc
>g729
>ulaw/alaw
>
> No GSM!
>
> How odd is that, given that it's a GSM mobile phone...
>
> Anyway, my quest for the ultimate "one handset" solution is getting
> closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor
> Granite it might have half a chance of working outside the room with the
> access point, however ...
>
> Anyone tried the Plantronics Voyager 510 bluetooth headsets which
> regsiters to both a mobile phone and their own base unit (which
> presumably has a USB sound device)
>
> as in:
>
> https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click
>
> I'm not a fan of soft-phones, and not sure I want to have a borg implant
> on when I'm not driving, but ...
>
> Oh well... Back to the grind!
>
> Gordon
>
> ___

I think that's pointless. Why do you need a USB audio device? You can
pair it to the computer directly and use it with any soft phone.

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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Baji Panchumarti

  On 10/31/07, Doug Lytle  wrote:

> Excellent!
>
> Thank you both!
>
> Doug
>

 don't forget that line of code will disappear the next time
 you upgrade your * addons, unless the change makes
 it into the official code base.

--

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Re: [asterisk-users] PRI commands missing...

2007-10-31 Thread Tzafrir Cohen

On Wed, Oct 31, 2007 at 10:46:42AM -0600, Carlos Chavez wrote:
> On Wed, 2007-10-31 at 09:41 -0600, Anthony Francis wrote:
> > This also happens if zaptel fails to load. Check your messages file.
> > 
> > John covici wrote:
> > > Well, this happened to me one time when I forgot to compile the pri
> > > library before the asterisk!  Could you have done that?
> > >
>   Asterisk is working with the second span with Unicall and that would
> not be possible unless Libpri and Zaptel are already loaded.

chan_unicall uses zaptel (the kernel interface) but not chan_zap.so (and
not libpri, IIRC).

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

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Re: [asterisk-users] (no subject)

2007-10-31 Thread Jim Houser

  We have used the Grandstream GPX2000, HT503 and GXW4104 gateways.  Quality
is in all cases are on the lower end.  The quality I refer to is buggy
software and poor call quality.  I have been involved with Telecom since the
early 80s and dealt with a lot of phone systems.  The Grandstream phones
just plain feel cheap.  Real "Walmart" quality, not professional business
class equipment.

  The phone functioned ok and was super easy to setup but complaints of echo
and poor volume levels were common.  They may be better as we have not used
them in over 6 months.

  We have recently used their gateways due to good pricing and their
economics fit our solution base well but ran into issues with them.  I
believe their gateways will get improved as both are new and on early
firmware releases.  However, we got upset with poor support.  Either no call
back at all or a useless email a day later with little to no information to
help solve our issue.  In Grandstream's defense it may be we are just too
small to matter and that's ok.  

  We prefer to go elsewhere and deliver product that when the average user
picks it up to talk on it they say "this is quality stuff".  Asterisk is as
talented as the firm that programs it BUT the phone is crucial in the end
user's system satisfaction.   Regardless of what you put in the back room
the phone IS the device that sets the impression to your client if you are
delivering a quality solution.

   We would do Cisco because it is high quality but we don't care to fight
with the configuration or licensing issues.  We would do Polycom, and
probably will, but have not had the time to jump to through the hoops needed
to acquire good enough pricing to make money selling them.  We feel Aastra
is a good compromise in delivering quality product to make the customer
happy with their decision while still making us to make some sort of small
profit for our time.  It's solid and provides a quality feel and function. 

  This said, Grandstream is not junk and this is not meant to be a
Grandstream rant.  I would like to apologize if I jumped in too quick
sounding that way.  Grandstream is just the lower end of quality and should
be deployed in applications where the client is willing to accept that.
That's not our marketplace.  If you want easy to configure, low cost, slam
dunk Asterisk deployments then Grandstream works.  But the end result will
not be as good if you build a system with Cisco, Polycom, Snom, or  Aastra.
We've even tested Avaya 46XX phones on Asterisk.  They sound GREAT!
Probably one of the best.  We just can't get Asterisk to light the messaging
waiting light on the phone.  Arrggg!

  You need to decide what your marketplace offering is and what your clients
are willing to accept.  If call quality is the most important then our
testing shows nobody beats Polycom or Avaya.  Someday we are going to beat
the Avaya message waiting light issue.  If quality of deskset feel is the
most important factor them Avaya and Cisco stand out as best.  We will not
put configuration into a factor simply because the customer uses the tool we
are expected to configure it to their needs.  We won't sell them any device
based on it being "easier" for us to configure.

  I would like to hear what people say about Snom as their sets look very
nice.  

Sorry for the novel, all I really wanted to express is Grandstream is cheap,
look before you jump.
Good luck on your decision...
Jim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Wednesday, October 31, 2007 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)

What is the issue with the Grandstream?  We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be pretty
good.  The Polycom work well, but they seem to die after about a year or so.
We bought 20 of them about 2 years ago and 7 of them have died or had
buttons stop working so we had to replace them.  I haven't had a single
Cisco do that and we have probably 100 of them.

Jim Houser wrote:
> We agree with Drew and no longer use Grandstream.   We have used a few
> Polycom, (best voice quality, hardest to configure).  I have heard 
> good things about Snom but never used them.  We standardized on 
> Aastra.  Good build, sound quality, and feature set.  Easy to 
> configure or upgrade and good pricing.  If you try Snom please share 
> your thoughts.  At present we are sticking with Aastra due to good results
and user feedback.
> 
> Jim
> 
> [EMAIL PROTECTED] wrote:
>> Hi all,
>>
>> We have a client that needs to setup about 80 desk phones (about 50 
>> in one location and about another 30 in 5 different locations). Which 
>> brand/model would you recommend. We were personally thinking in 
>> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard 
>> great things about them. However, having no real experience with t

Re: [asterisk-users] queues without 302 redirects?

2007-10-31 Thread Louis-David Mitterrand

On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote:
> Hi,
> 
> Using 1.4.13 is it possible to ignore 302 redirects from sip devices 
> belonging to a queue?
> 
> For a queue that rings the whole office it doesn't seem very useful to 
> obey a redirect programmed on a phone.
> 
> It seems this was the default behaviour in 1.2.

For the record and google the answer is the 'i' option in Queue().

Thanks again to Strom_M on #asterisk!

god I love IRC...

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[asterisk-users] queues without 302 redirects?

2007-10-31 Thread Louis-David Mitterrand

Hi,

Using 1.4.13 is it possible to ignore 302 redirects from sip devices 
belonging to a queue?

For a queue that rings the whole office it doesn't seem very useful to 
obey a redirect programmed on a phone.

It seems this was the default behaviour in 1.2.

Thanks,

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Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls

2007-10-31 Thread Mojo with Horan & Company, LLC

Barry D. Hassler wrote:
> I've tried to find other threads with this same topic, but haven't 
> found any... Apologies if this already being discussed
>
> Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4.
>
> Having an issue with (I think) parked calls. We tend to park calls, 
> but we're often not able to pick them back up, or the other party says 
> they get dropped, etc. There doesn't seem to be a specific pattern 
> that I've discovered so far. I had this happen to me personally this 
> morning -- receptionist parked a call for me on extension 7001, but 
> when I dialed 7001, just got dead air. I could see in asterisk that 
> the call was indeed parked though, and after calling the person back, 
> he reported he was just hearing the lovely on-hold music.
>
> Is there a known issue (and even better, a fix) for this situation? 
> Any other information I can provide I'll do so!
What kind of phones are you using? are they Zap or SIP?

Can you provide a CLI output with any tips in it?



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[asterisk-users] SIP_INFO

2007-10-31 Thread Christophorus Laube

Hi list,

does anyone of you know wether asterisk can handle SIP_INFO on pure sip
calls? Is that something I have to handle in the extensions? Does
asterisk hand incoming SIP_INFO over to an already connected peer?
Thanks and regards,

Christophorus Laube


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[asterisk-users] Mobile phone codecs ...

2007-10-31 Thread Gordon Henderson


Not strictly asterisk related, however...

Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client 
which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?)

Anyway, in a fit of idleness, I thought I'd see what codecs it supports, 
as I couldn't find it in the manual...

And it supports:

   ilbc
   g729
   ulaw/alaw

No GSM!

How odd is that, given that it's a GSM mobile phone...

Anyway, my quest for the ultimate "one handset" solution is getting 
closer. If Wi-Fi weren't so rubbish and my house not made of Dartmoor 
Granite it might have half a chance of working outside the room with the 
access point, however ...

Anyone tried the Plantronics Voyager 510 bluetooth headsets which 
regsiters to both a mobile phone and their own base unit (which 
presumably has a USB sound device)

as in:

https://www.ukheadsets.co.uk/thc-plantronics-voyager-510-usb-dongle-rn-422-action-show_detail-show_products_mode-cat_click

I'm not a fan of soft-phones, and not sure I want to have a borg implant 
on when I'm not driving, but ...

Oh well... Back to the grind!

Gordon

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Re: [asterisk-users] PRI commands missing...

2007-10-31 Thread Carlos Chavez

On Wed, 2007-10-31 at 09:41 -0600, Anthony Francis wrote:
> This also happens if zaptel fails to load. Check your messages file.
> 
> John covici wrote:
> > Well, this happened to me one time when I forgot to compile the pri
> > library before the asterisk!  Could you have done that?
> >
Asterisk is working with the second span with Unicall and that would
not be possible unless Libpri and Zaptel are already loaded.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] issues with downloads.digium.com

2007-10-31 Thread Carlos Chavez

On Wed, 2007-10-31 at 15:26 +0200, Tzafrir Cohen wrote:
> On a slightly different matter: 
> http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
> 1.4.1 .
> 

Yes, I noticed that too and was wondering if it is just because they
have not updated the site or if there is a problem with the newest
versions and they do not want people to download them.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] T.38 Faxing and Asterisk

2007-10-31 Thread Paul Bryson

Nasir Iqbal wrote:
> Hi,
> 
> 
> Have you tried Callweaver http://www.callweaver.org

I was really hoping to be able to use Trixbox to do this and it's a 
pretty complete solution by itself.  Unfortunately that requires Asterisk.

It appears that there is no way to get Asterisk, or anything on the 
Asterisk box, to act as a T.38 endpoint.  This appears to be the result 
of a licensing issue with SpanDSP.
http://www.voip-info.org/wiki/view/T.38

That's a real shame as T.38 termination support is one of the last big 
pieces for us to make Asterisk a seamless solution.

Paul Bryson

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Re: [asterisk-users] (no subject)

2007-10-31 Thread Peder @ NetworkOblivion

What is the issue with the Grandstream?  We are getting tired of Cisco 
issues, so we have started looking at Grandstream and they seem to be 
pretty good.  The Polycom work well, but they seem to die after about a 
year or so.  We bought 20 of them about 2 years ago and 7 of them have 
died or had buttons stop working so we had to replace them.  I haven't 
had a single Cisco do that and we have probably 100 of them.

Jim Houser wrote:
> We agree with Drew and no longer use Grandstream.   We have used a few
> Polycom, (best voice quality, hardest to configure).  I have heard good
> things about Snom but never used them.  We standardized on Aastra.  Good
> build, sound quality, and feature set.  Easy to configure or upgrade and
> good pricing.  If you try Snom please share your thoughts.  At present we
> are sticking with Aastra due to good results and user feedback.
> 
> Jim
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
> Sent: Wednesday, October 31, 2007 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] (no subject)
> 
> [EMAIL PROTECTED] wrote:
>> Hi all,
>>
>> We have a client that needs to setup about 80 desk phones (about 50 in 
>> one location and about another 30 in 5 different locations). Which 
>> brand/model would you recommend. We were personally thinking in 
>> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard 
>> great things about them. However, having no real experience with them 
>> makes it hard in recommending one to our customer. The only experience 
>> we've had is a very frustrating one trying to load the IP software on 
>> a Cisco 7970G and so we assume that if we have to go through that for 
>> all 80 phones, we'll probably commit suicide :)
>>
>> Thanks
>>   
> 
> We have used Cisco and Aastra, can't comment on Polycom or Snom.
> 
> I cannot recommend Cisco, good sound quality but that's it. Ridiculously
> overpriced, too few usable features, incredibly awkward to manage.
> Aastra have good sound quality, reasonable price, configs are plain text and
> not to hard to work with. We have the 9133i as our basic phone and 480i in
> the Call Centre for the soft buttons. Both can be fed from the same config
> templates.
> We used to use Grandstream but quality and support issues have driven us
> away.
> 
> regards,
> 
> Drew
> 
> --
> Drew Gibson
> 
> Systems Administrator
> OANDA Corporation
> www.oanda.com
> 
> 
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> 


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Re: [asterisk-users] (no subject)

2007-10-31 Thread Jim Houser

We agree with Drew and no longer use Grandstream.   We have used a few
Polycom, (best voice quality, hardest to configure).  I have heard good
things about Snom but never used them.  We standardized on Aastra.  Good
build, sound quality, and feature set.  Easy to configure or upgrade and
good pricing.  If you try Snom please share your thoughts.  At present we
are sticking with Aastra due to good results and user feedback.

Jim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Wednesday, October 31, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)

[EMAIL PROTECTED] wrote:
> Hi all,
>
> We have a client that needs to setup about 80 desk phones (about 50 in 
> one location and about another 30 in 5 different locations). Which 
> brand/model would you recommend. We were personally thinking in 
> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard 
> great things about them. However, having no real experience with them 
> makes it hard in recommending one to our customer. The only experience 
> we've had is a very frustrating one trying to load the IP software on 
> a Cisco 7970G and so we assume that if we have to go through that for 
> all 80 phones, we'll probably commit suicide :)
>
> Thanks
>   

We have used Cisco and Aastra, can't comment on Polycom or Snom.

I cannot recommend Cisco, good sound quality but that's it. Ridiculously
overpriced, too few usable features, incredibly awkward to manage.
Aastra have good sound quality, reasonable price, configs are plain text and
not to hard to work with. We have the 9133i as our basic phone and 480i in
the Call Centre for the soft buttons. Both can be fed from the same config
templates.
We used to use Grandstream but quality and support issues have driven us
away.

regards,

Drew

--
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] (no subject)

2007-10-31 Thread Drew Gibson

[EMAIL PROTECTED] wrote:
> Hi all,
>
> We have a client that needs to setup about 80 desk phones (about 50  
> in one location and about another 30 in 5 different locations). Which  
> brand/model would you recommend. We were personally thinking in  
> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard  
> great things about them. However, having no real experience with them  
> makes it hard in recommending one to our customer. The only  
> experience we've had is a very frustrating one trying to load the IP  
> software on a Cisco 7970G and so we assume that if we have to go  
> through that for all 80 phones, we'll probably commit suicide :)
>
> Thanks
>   

We have used Cisco and Aastra, can't comment on Polycom or Snom.

I cannot recommend Cisco, good sound quality but that's it. Ridiculously 
overpriced, too few usable features, incredibly awkward to manage.
Aastra have good sound quality, reasonable price, configs are plain text 
and not to hard to work with. We have the 9133i as our basic phone and 
480i in the Call Centre for the soft buttons. Both can be fed from the 
same config templates.
We used to use Grandstream but quality and support issues have driven us 
away.

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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Re: [asterisk-users] Size of Exten when using IAX

2007-10-31 Thread Arjan Kroon | Mobillion

If I look at the console (with verbosity on 3) I see that also the last
4 characters are lost.

I never heard of 'wireshark on the wire' I'll try this.

Is IAXVARS also supported on asterisk 1.0.0 ?

--

Arjan Kroon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: dinsdag 30 oktober 2007 15:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Size of Exten when using IAX

On Tuesday 30 October 2007 08:40:51 Arjan Kroon | Mobillion wrote:
> We are use IAX protocol between two asterisk servers.
>
> Now we send information through this protocol by using EXTEN
>
>
>
> We see that the variable EXTEN only holds 66 characters.
>
> If we set a value larger then 66 characters, for example 70
characters.
>
> The last 4 characters are cut off.
>
>
>
> Is there a way to increase this variable?

You're going to have to provide more information for us to help you.
There are numerous places where the extension string could be getting
truncated, so you'll have to look some more:

1) On the console, with verbose set to 3 or higher, when the dialplan is
executed, are you showing all of the numbers?
2) If you run wireshark on the wire, does the IAX2 packet show all of
the
numbers in the CALLED_NUMBER IE?

Also, you should know that in trunk, there is a much better way of
transmitting independent bits of data about the call, called IAXVARS.
We're presently looking at abstracting this into something a bit more
protocol independent, but that's the way it is presently.

-- 
Tilghman

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Re: [asterisk-users] PRI commands missing...

2007-10-31 Thread Anthony Francis

This also happens if zaptel fails to load. Check your messages file.

John covici wrote:
> Well, this happened to me one time when I forgot to compile the pri
> library before the asterisk!  Could you have done that?
>
> on Wednesday 10/31/2007 Tzafrir Cohen([EMAIL PROTECTED]) wrote
>  > On Wed, Oct 31, 2007 at 12:06:25AM -0600, Carlos Chavez wrote:
>  > >  I have an Asterisk server running Elastix but patched to use 
> Unicall. 
>  > > Everything seems to be working fine and the TE220 card is up and running 
> with
>  > > port 1 configured as PRI and port 2 as MFC/R2.  We can already send and
>  > > receive calls on port two but we cannot on port one.  That is when we 
> noticed
>  > > that there are no PRI commands available on the Asterisk CLI.  We cannot 
> use
>  > > PRI DEBUG SPAN to determine why port 1 is not receiving or sending calls.
>  > > 
>  > >  Why would this commands be missing?  
>  > 
>  > I wonder how those two should interact. The first thing chan_zap tries
>  > to do is to open all of its spans. Maybe it has failed there?
>  > 
>  > Try playing with [trunkgroups] to explicitly tell it to only touch the
>  > Zaptel spans that are PRI.
>  > 
>  > -- 
>  >Tzafrir Cohen   
>  > icq#16849755  jabber:[EMAIL PROTECTED]
>  > +972-50-7952406   mailto:[EMAIL PROTECTED]   
>  > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>  > 
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>   

-- 
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]


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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Doug Lytle

Sean Bright wrote:
> Find this line:
>
> if (mysql_real_connect(mysql, dbhost, dbuser...

Excellent!

Thank you both!

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] Best cheap card to use for home Asterisk system???

2007-10-31 Thread Darrick Hartman (lists)

Tim Reimers wrote:
> I have a single phone line (happens to be Charter Communications VOIP, 
> but I have their ATA and they’ve connected to red/green pair in the 
> house wiring)

Ok. so they've installed an ATA which connects your analog phones to 
their VoIP (perhaps SIP) service.

> What I’d like to do is this:
> 
> Get some low-end but reliable card/external adapter which would connect 
> to their ATA and tie into Asterisk to take calls and faxes

OK.  Since we've established above that Charter's service is VoIP 
converted to analog, AND since Asterisk isn't really designed to work 
with fax over IP it is safe to say that it's not worth the effort to 
attempt to get this to work.  I have relatives who have Time Warner's 
offering and even a stand alone fax machine will not work reliably over 
their "internet phone" service.  Hell the audio quality is crap most of 
the time.

> I’m assuming this should be something with one FXO and one FXS port to 
> connect the incoming line to and to connect the red/green wiring in the 
> house to.

I'm not sure if you're familiar with the Canadian television show that 
is popular on PBS in the US, but this sounds alot like the guy on the 
Red Green Show using duct tape to fix things.  If you really want to use 
Asterisk, you'd be better off getting an account with a SIP provider and 
using an FXS adapter to feed "line 2" on your phones similar to what 
Charter is doing with line 1.  Linksys makes a decent adapter which 
would suit this purpose.

Good luck!

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] Asterisk 1.4.13 -- issue with parked calls

2007-10-31 Thread Doug Lytle

Barry D. Hassler wrote:
> Is there a known issue (and even better, a fix) for this situation? 
> Any other information I can provide I'll do so!


How are the calls getting parked?

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Sean Bright

Find this line:

if (mysql_real_connect(mysql, dbhost, dbuser...

Add this before that line:

int timeout = 10; /* 10 second timeout */
mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, (const char *) &timeout);

And recompile.

On 10/31/07, Doug Lytle <[EMAIL PROTECTED]> wrote:
>
> Tilghman Lesher wrote:
> > The key would be adding this line at the appropriate point:
> > mysql_options(&mysql, MYSQL_OPT_CONNECT_TIMEOUT, &timeout)
> > where timeout is an integer.  Remember that it needs to be set
> > BEFORE the connection.
> >
> >
>
> Anybody like to give more detailed instructions for those of use not
> instructed in C?
>
> Thanks!
>
> Doug
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
>
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[asterisk-users] Asterisk 1.4.13 -- issue with parked calls

2007-10-31 Thread Barry D. Hassler

I've tried to find other threads with this same topic, but haven't found
any... Apologies if this already being discussed

Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4.

Having an issue with (I think) parked calls. We tend to park calls, but
we're often not able to pick them back up, or the other party says they get
dropped, etc. There doesn't seem to be a specific pattern that I've
discovered so far. I had this happen to me personally this morning --
receptionist parked a call for me on extension 7001, but when I dialed 7001,
just got dead air. I could see in asterisk that the call was indeed parked
though, and after calling the person back, he reported he was just hearing
the lovely on-hold music.

Is there a known issue (and even better, a fix) for this situation? Any
other information I can provide I'll do so!




-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Doug Lytle

Tilghman Lesher wrote:
> The key would be adding this line at the appropriate point:
> mysql_options(&mysql, MYSQL_OPT_CONNECT_TIMEOUT, &timeout)
> where timeout is an integer.  Remember that it needs to be set
> BEFORE the connection.
>
>   

Anybody like to give more detailed instructions for those of use not 
instructed in C?

Thanks!

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] SIP phone recommendation (used to be: no subject)

2007-10-31 Thread Barry D. Hassler

I'd go with Polycom all the way. We have a number of different types of
phones in use, or that we've worked with, including Grandstream, SIpura and
Atacom, and the quality difference with the Polycom phones is astounding.

On 10/29/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> My apologies to the list for not having entered a subject line in the
> email.
>
> Thanks
>
> On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote:
>
> > Hi all,
> >
> > We have a client that needs to setup about 80 desk phones (about 50
> > in one location and about another 30 in 5 different locations). Which
> > brand/model would you recommend. We were personally thinking in
> > recommending either Cisco, Aastra, Polycom, or Snom, for we've heard
> > great things about them. However, having no real experience with them
> > makes it hard in recommending one to our customer. The only
> > experience we've had is a very frustrating one trying to load the IP
> > software on a Cisco 7970G and so we assume that if we have to go
> > through that for all 80 phones, we'll probably commit suicide :)
> >
> > Thanks
> >
> >
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-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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[asterisk-users] Best cheap card to use for home Asterisk system???

2007-10-31 Thread Tim Reimers

Hi all -

 

I'm building an Asterisk system (Trix2.2) for the house-

 

I'd like to do the following things:

 

I have a single phone line (happens to be Charter Communications VOIP,
but I have their ATA and they've connected to red/green pair in the
house wiring)

 

What I'd like to do is this:

 

Get some low-end but reliable card/external adapter which would connect
to their ATA and tie into Asterisk to take calls and faxes

I'm assuming this should be something with one FXO and one FXS port to
connect the incoming line to and to connect the red/green wiring in the
house to.

 

I don't mind if all the house phones ring at one time for the moment, as
line 2 on them are the Asterisk extensions.

 

 

Whatever I use must also have failover capability, such that when
Asterisk is not working right (server down completely OR just not
responding) 

then the unit fails over and cross connects and makes things work just
as is normally the case with Charter only.

 

 

Unfortunately, I don't have a budget of hundreds of dollars for a true
Digium multiport card -

 

I've already built out Asterisk and have a Cisco ATA supporting line 2
on a couple of cordless phones,

but I'd like to have the failover piece so that if * starts failing, the
home phones still work..

 

Thanks, Tim

 

 

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Re: [asterisk-users] issues with downloads.digium.com

2007-10-31 Thread Tzafrir Cohen

On a slightly different matter: 
http://asterisk.org/downloads still points to zaptel 1.4.5.1 and libpri
1.4.1 .

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

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Re: [asterisk-users] Astribank + AsteriskNOW

2007-10-31 Thread Tzafrir Cohen

On Wed, Oct 31, 2007 at 10:54:22AM -0200, Guilherme Loch Waltrick Góes wrote:
> Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in
> mantis seems to be closed, but I cannot find "fxload" or "lsusb" to do some
> debugging.

Please use the latest beta: 6.5:

http://www.rpath.org/rbuilder/project/asterisk/

Those issues, along with a number of smaller issues, have been resolved
there.

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

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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Tilghman Lesher

On Tuesday 30 October 2007 18:19:33 Douglas Garstang wrote:
> Anyone know if the MySQL() application has a configurable timeout?

It does not.

> If it tries to connect to a bogus IP, it's timeout seems to be a few
> minutes. I'd like to cut it down to a few seconds.

The key would be adding this line at the appropriate point:
mysql_options(&mysql, MYSQL_OPT_CONNECT_TIMEOUT, &timeout)
where timeout is an integer.  Remember that it needs to be set
BEFORE the connection.

-- 
Tilghman

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Re: [asterisk-users] PRI over T1 calls dropping, cause 100

2007-10-31 Thread Michelle Dupuis

The T1 was setup as tie line, not a trunk.   The Bell guy tried setting up
the line 2 ways:

1.  As a trunk.  This did not work because:
  a)  When he typed in the access code for the trunk on a phone set (and
then any numbers), the call never appeared on the Asterisk side.
  b)  The Bell guy said that unless Asterisk was generating a dialtone, a
trunk would not work.
  (I struggled to understand these explanations...but figured I must be
missing something)
2.  As a tie line.  This sort of worked because
  a)  When he typed the access code for the tie line on a phone set, he got
a second dial tone.
  b)  When he dialed any digits thereafter, the call was handed across the
T1 and I saw it on the asterisk side.

Can you give me any specifics (or a link) on how the Meridian side should be
configured?  

Thanks,
MD

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Andres
> Sent: Tuesday, October 30, 2007 11:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PRI over T1 calls dropping, cause 100
> 
> 
> >-- Processing IE 24 (cs0, Channel Identification)
> >-- Processing IE 28 (cs0, Facility)
> >Handle Q.932 ROSE Invoke component
> >-- Processing IE 108 (cs0, Calling Party Number)
> >
> >
> The Meridian is trying to Invoke the Remote Operations 
> Service Element (ROSE).  That is used to support interactive 
> applications.  My guess is that Meridian thinks its talking 
> to another Meridian and its trying to startup some 
> application.  That is not going to play well with Asterisk.  
> You need to see how to disable that, or configure a plain 
> trunk without any fancy stuff at the Meridian side.
> 
> Andres.
> 
> 
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Re: [asterisk-users] Correct voltages but no dial tone on TDM2400P

2007-10-31 Thread Tilghman Lesher

On Tuesday 30 October 2007 18:22:21 Alex R Green wrote:
> The TDM2400P card has three green FXS modules close to the 50pin
> connector and three red FXO modules at the rear. The card was installed
> in the system prior to loading Trixbox. I think the card is working:
> zaptel.conf was set up correctly with the first 12 channels fxsks and
> the last 12 channels fxo kewlstart.

Your signalling is backwards.  The first 12 channels are fxs modules,
but they are signalled with fxo signalling, so fxoks=1-12, and the last
12 are fxo modules, but they are signalled with fxs signalling, so
fxsks=13-24.  Ditto for zapata.conf.

-- 
Tilghman

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[asterisk-users] Astribank + AsteriskNOW

2007-10-31 Thread Guilherme Loch Waltrick Góes

Does anyone got Astribank working with AsteriskNOW beta 6 ? The bug in
mantis seems to be closed, but I cannot find "fxload" or "lsusb" to do some
debugging.

-- 
Guilherme Loch Góes

MSN:[EMAIL PROTECTED]
(48) 99115299
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Re: [asterisk-users] flooded by "Maximum trunk data space exceeded" messages

2007-10-31 Thread Arun Kumar

try to reduce number of calls on trunk or create multiple trunks.

On 10/31/07, Louis-David Mitterrand <[EMAIL PROTECTED]>
wrote:
>
> Hi,
>
> Using 1.4.13 and trunking a single iax channel to a similar box my
> asterisk console is flooded with:
>
> [Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data
> space exceeded to xx.xx.xx.xx:4569
>
> Known issue?
>
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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Baji Panchumarti

  On 10/31/07, Douglas Garstang wrote:

> I guess... it shouldn't be too hard to find the time out value
> in the source and change it

 I couldn't find any timeout related parameter in

app_addon_sql_mysql.c

 You may find a default value in one of the header files.

 I am wondering if it wouldn't be easier to try and detect
 the existence of the DB host via System().

 -baji.

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Re: [asterisk-users] G.729 required for IP<--->TDM<--->IP

2007-10-31 Thread Gordon Henderson

On Tue, 30 Oct 2007, satish patel wrote:

> Dear all
>
> I have already post this question but i need more input for 
> this setup
>
> [IPphone]--[Asterisk]E1---[Avaya]---[ip_Extention]
>
> Asterisk - codec (G.711/ulaw)
> Avaya - codec ( G.711/ulaw)
>
> Now I need G.729 on my asterisk side and i have put G.729 codec setting 
> on my IP phone and when i make call from asterisk to Avaya Extention i 
> got error
>
> translator not in path
>
> so i need to get license of g.729 on asterisk for transcoder or it will 
> work wothout translator ???
>
> My question is :-- Is there Required G.729 (License) on Asterisk Or Not 
> ???

You can purchase them from Digium:

http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5

$10 each.

Install one license for each simultaneous g792 call you expect to take on 
the asterisk box and off you go.

There are free versions of g729 avalable, but if your country is 
compatable with the various (US) patent laws then you ought to pay the 
license fee to stay legal.

Gordon

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Re: [asterisk-users] Faxing with Asterisk & SpanDSP [Was Fax Problems with SpanDSP]

2007-10-31 Thread Gordon Henderson

On Wed, 31 Oct 2007, Alan Lord wrote:

> Steve Underwood wrote:
>  ...
>> SpanDSP handles faxes within Asterisk, through the app_rxfax and
>> app_txfax applications. It handles faxes outside Asterisk when used with
>> iaxmodem (there is actually a copy inside the iaxmodem package).
>>
>> SpanDSP cannot be used by the standard distribution of Asterisk, as it
>> is GPL code. However, if you are using Asterisk within the restrictions
>> of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
>>
>> Regards,
>> Steve
>
> Thanks Steve,
>
> I was a bit lazy when I posted that question sorry - I just got a bit
> excited. A quick google and looking around the voip-info.org site
> pointed me to some interesting pages.
>
> Could anyone who has experience confirm if:
>
> * this will work with an x100p card so I can have a single FXO line that
> can detect incoming faxes and/or voice calls and enable me to use it for
> outgoing voice and fax?

I've used it in exactly this mode with TDM400 cards. It worked mostly OK. 
You might need to fiddle with the gains in the zapata.conf file. (although 
I now steer people away from this way of doing it and use a separate ATA 
if they have a real fax machine, or an external fax to email service)

> Also, I read somewhere that Asterisk should "never" be used with fax.
> IIRC correctly it was by the moderator on the Trixbox forum discussion
> about the trial of OSLEC. Yes, here:
> http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems
>
> "I really don't know how many times I need to say this but you should
> never run faxes through Asterisk. It was not designed to handle it.
> --
> Kerry Garrison
> trixbox Community Director"
>
> Is he right?

Dunno, but it works. FAX data is nothing special - however modems are very 
critical of jitter - you and me and tolerate a bit of packet loss, or the 
odd duplicate, etc. in an audio stream, a modem can't. (NO CARRIER :)

So your asterisk box shouldn't have jitter internally, but if it's running 
cpu intensive applications, it might have, I guess...

A lot of people have success with running iaxmodem - which is just spanDSP 
connected to some fancy code to understand AT commands, and they then 
plumb this into hylafax running on a separate server over ethernet.. I've 
tried ATAs over ethernet to send/receive faxes to real fax machines with 
good results too. (Make sure the codec is G711 though, as nothing else 
will work)

Gordon

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Re: [asterisk-users] segfault - asterisk crash and restart

2007-10-31 Thread Atis Lezdins

Rilawich Ango wrote:
> Hi all,
> 
>   Recently, I have upgraded the asterisk as following.
> asterisk-1.4.13
> asterisk-addon-1.4.4
> libpri-1.4.1
> zaptel-1.4.5.1
> Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh
>   After upgrade, the server get segfault randomly and asterisk crash
> and restart itself.  I got 2 core dumps of the segfault.  Based on the
> core dump, we can't figure out the root cause to the problem as the
> content of the core dump is not the same.  We have no idea what the
> problem is.  Anyone can give me some advices.
> 
> --core dump 1--
> (gdb) bt full
> #0  0x0037e806e1f3 in _int_free () from /lib64/libc.so.6
> No symbol table info available.
> #1  0x0037e8071fac in free () from /lib64/libc.so.6
> No symbol table info available.
> #2  0x0046b7b7 in ast_frame_free (fr=0x1b9da4b0, cache=0)
> at frame.c:369
> No locals.
> #3  0x2aaab1173573 in mixmonitor_thread (obj=0x1bb08220)
>from /usr/lib/asterisk/modules/app_mixmonitor.so

This is clearly mixmonitor-related. I suggest you to look for similar
mixmonitor bugs in digium's mantis - if there's none, create and attach
this backtrace.

[snip]

> --core dump 2--
> 
> Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
> Program terminated with signal 11, Segmentation fault.
> #0  0x0044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69
> 69if (name[0] == '_') {
> (gdb) bt full
> #0  0x0044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69
>   name = 0x10f1d58b0 
> #1  0x0049948f in pbx_builtin_setvar_helper (chan=0xf460320,
> name=0x2aaabf53cbf7 "DIALSTATUS",
> value=0x417a0690 "BUSY") at pbx.c:5825
>   newvariable = (struct ast_var_t *) 0x10f1d58a0
>   headp = (struct varshead *) 0xf460880
>   nametail = 0x2aaabf53cbf7 "DIALSTATUS"
>   __PRETTY_FUNCTION__ = "pbx_builtin_setvar_helper"

Great, this confirms that i'm not the only one having this problem. Can
you please add this to http://bugs.digium.com/view.php?id=10923

As from my knowledge - this will happen often on 1.4.13.. The safe
version i'm using is 1.4.10, but 1.4.12.1 already have this problem..

Regards,
Atis


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[asterisk-users] flooded by "Maximum trunk data space exceeded" messages

2007-10-31 Thread Louis-David Mitterrand

Hi,

Using 1.4.13 and trunking a single iax channel to a similar box my 
asterisk console is flooded with:

[Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space 
exceeded to xx.xx.xx.xx:4569

Known issue?

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Re: [asterisk-users] MySQL() timeout

2007-10-31 Thread Doug Lytle

Douglas Garstang wrote:
> I guess... it shouldn't be too hard to find the time out value in the 
> source and change it
>

If you find the line, please let me know where. 

Doug

-- 
Ben Franklin quote:

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Safety, deserve neither Liberty nor Safety."



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[asterisk-users] Faxing with Asterisk & SpanDSP [Was Fax Problems with SpanDSP]

2007-10-31 Thread Alan Lord

Steve Underwood wrote:
 ...
> SpanDSP handles faxes within Asterisk, through the app_rxfax and 
> app_txfax applications. It handles faxes outside Asterisk when used with 
> iaxmodem (there is actually a copy inside the iaxmodem package).
> 
> SpanDSP cannot be used by the standard distribution of Asterisk, as it 
> is GPL code. However, if you are using Asterisk within the restrictions 
> of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
> 
> Regards,
> Steve

Thanks Steve,

I was a bit lazy when I posted that question sorry - I just got a bit 
excited. A quick google and looking around the voip-info.org site 
pointed me to some interesting pages.

Could anyone who has experience confirm if:

* this will work with an x100p card so I can have a single FXO line that 
can detect incoming faxes and/or voice calls and enable me to use it for 
outgoing voice and fax?

Also, I read somewhere that Asterisk should "never" be used with fax. 
IIRC correctly it was by the moderator on the Trixbox forum discussion 
about the trial of OSLEC. Yes, here: 
http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems

"I really don't know how many times I need to say this but you should 
never run faxes through Asterisk. It was not designed to handle it.
--
Kerry Garrison
trixbox Community Director"

Is he right?

The question that prompted this reply was about how to turn off echo 
cancellation for fax traffic. Is this achievable?

Many thanks

Alan

-- 
The way out is open!
http://www.theopensourcerer.com

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