[Asterisk-Users] 1.2.7.1 on FC5 won't make install

[Cliff Savage]

The make seems to go okay.

[EMAIL PROTECTED] asterisk-1.2.7.1]# uname -a
Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux


mkdir -p /var/lib/asterisk/sounds/digits
mkdir -p /var/lib/asterisk/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if grep -q "^%`basename $x`%" sounds.txt; then \
install -m 644 $x /var/lib/asterisk/sounds/digits ; \
else \
echo "No description for $x"; \
exit 1; \
fi; \
done
No description for sounds/digits/*.gsm
make: *** [datafiles] Error 1
 

All the folders in /var/lib/asterisk/sounds are full
of sounds except digits and priv-caller-intros.
Those 2 folders are empty.

Manually copy them in and recompile maybe???



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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[William M Conlon]

it's the cable that is 802.3af, not the phone.
On Apr 21, 2006, at 5:01 PM, Steven Ringwald wrote:


William M Conlon wrote:
Beware.  Although the Polycom 501s will sink power over ethernet  
(that is they are powered by a cable pair within a cable that  
resembles ethernet), they are NOT IEEE 802.3af POE devices!   
They work on 12 volts (I think -- haven't measured it) instead of  
48VDC.  So don't expect to buy a POE source and expect the phone  
to receive power just by plugging in a patch cable.  They must be  
powered by the Polycom voltage sources.


Nevertheless, here's what works for me:

Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch  
cable :: Polycom 501


Some notes:
1.  The Polycom injector cable should be plugged into a POE port  
on the switch (the Netgear FS108 switch has both powered and  
unpowered ports), or the Polycom injector will not source power.

2.  The Netgear FS108 is NOT sourcing power.
3.  The patch cable is a 50-foot CAT5.
3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE  
phone, IMHO.  Caveat emptor.


They are 802.3af, if you buy the correct cable (they two different  
types):


(From http://www.voip-info.org/wiki-Polycom+Phones)

(phones with the cables)
SoundPoint IP 501 (NA PSU)|2200-11531-001|$270
SoundPoint IP 501 (IEEE PoE)|2200-11531-025|$295

(just the cables)
NA PSU for 30x,50x,600 Qty 5|2200-07496-001|$35
IEEE PoE cable for 30x,50x|2200-11077-002|$35
Cisco PoE cable for 30x,50x|2200-11014-002|$35

I believe the reason the 30x and 50x's are like this is because the  
standard was still in flux when they were designed. They put the  
POE "brains" into the cable to make it easy to switch to whatever  
standard was decided upon. The 601 comes with built in POE  
(compatible with Cisco and IEEE).


Steve

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Bill

William M. Conlon, P.E., Ph.D.
To the Point
345 California Avenue Suite 2
Palo Alto, CA 94306
   vox:  650.327.2175 (direct)
   fax:  650.329.8335
mobile:  650.906.9929
e-mail:  mailto:[EMAIL PROTECTED]
   web:  http://www.tothept.com

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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Steven Ringwald]

William M Conlon wrote:
Beware.  Although the Polycom 501s will sink power over ethernet (that 
is they are powered by a cable pair within a cable that resembles 
ethernet), they are NOT IEEE 802.3af POE devices!  They work on 12 
volts (I think -- haven't measured it) instead of 48VDC.  So don't 
expect to buy a POE source and expect the phone to receive power just 
by plugging in a patch cable.  They must be powered by the Polycom 
voltage sources.


Nevertheless, here's what works for me:

Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch cable 
:: Polycom 501


Some notes:
1.  The Polycom injector cable should be plugged into a POE port on 
the switch (the Netgear FS108 switch has both powered and unpowered 
ports), or the Polycom injector will not source power.

2.  The Netgear FS108 is NOT sourcing power.
3.  The patch cable is a 50-foot CAT5.
3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE phone, 
IMHO.  Caveat emptor. 


They are 802.3af, if you buy the correct cable (they two different types):

(From http://www.voip-info.org/wiki-Polycom+Phones)

(phones with the cables)
SoundPoint IP 501 (NA PSU)|2200-11531-001|$270
SoundPoint IP 501 (IEEE PoE)|2200-11531-025|$295

(just the cables)
NA PSU for 30x,50x,600 Qty 5|2200-07496-001|$35
IEEE PoE cable for 30x,50x|2200-11077-002|$35
Cisco PoE cable for 30x,50x|2200-11014-002|$35

I believe the reason the 30x and 50x's are like this is because the 
standard was still in flux when they were designed. They put the POE 
"brains" into the cable to make it easy to switch to whatever standard 
was decided upon. The 601 comes with built in POE (compatible with Cisco 
and IEEE).


Steve

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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Steven Ringwald]

Steve Kennedy wrote:

On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:

  

D-link has a nice one, optional 5 year warranty on some of the
commercial stuff



Though beware, some of the D-Link ones only have half the ports with
PoE.
  


Actually, as far as I know, only one of the D-Link POE switches is like 
that, the DES-1316 has 16 ethernet ports, with 8 of them POE. The 1516 
has 26 POE ports (with two of them gig).


Steve


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RE: [Asterisk-Users] Unicall MFRC2 Problems with BrT.

[Anton Krall]

Moises, how can I find out which version Im running, on Steves ftp all say
0.0.3 or the date also says the same date.


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Moises Silva
|Sent: Friday, April 21, 2006 9:43 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
|
|A couple of weeks ago, libmfcr2 has a small error in the tone 
|signaling for the call setup, that was fixed 2 weeks ago or 
|so, please, wich version of libmfcr2 are you using? if you 
|dont know try upgrading to the latest version. Im pretty much 
|sure that you have the very same problem we had.
|
|Regards
|
|On 4/21/06, Jefferson Carvalho <[EMAIL PROTECTED]> wrote:
|> Hello All,
|>
|> I'm facing problems with Unicall on this scenario :
|>
|> CentOS 4.3 - Running on x86_64
|> Asterisk 1.2.7.1
|> Zaptel 1.2.5
|>
|> When running zttool , shows all Spans OK.
|>
|> But I can't receive and make calls.
|>
|> I tried to change many parameters and still doesn't work.
|>
|> Any clues ?
|>
|> * unicall.conf
|>
|> [channels]
|>
|> language=br
|>
|> context=incoming-pstn
|> usecallerid=yes
|> hidecallerid=no
|> immediate=no
|> callwaitingcallerid=yes
|> threewaycalling=yes
|> transfer=yes
|> cancellforward=yes
|> callreturn=yes
|> echocancel=yes
|> echocancelwhenbridged=yes
|>
|> rxgain=0.0
|> txgain=0.0
|> faxdetect=both
|> loglevel=255
|> protocolclass=mfcr2
|> protocolvariant=br,20,4
|> protocolend=cpe
|> group=1
|> callgroup=1
|>
|> channel => 1-15
|> channel => 17-31
|> channel => 32-46
|> channel => 48-62
|> channel => 63-77
|> channel => 94-108
|> channel => 110-124
|>
|> * zaptel.conf *
|>
|> loadzone=br
|> defaultzone=br
|>
|>
|> span=1,1,0,cas,hdb3
|> cas=1-15:1101
|> cas=17-31:1101
|>
|> span=2,0,0,cas,hdb3
|> cas=32-46:1101
|> cas=48-62:1101
|>
|>
|> span=3,0,0,cas,hdb3
|> cas=63-77:1101
|> cas=79-93:1101
|>
|> span=4,0,0,cas,hdb3
|> cas=94-108:1101
|> cas=110-124:1101
|>
|>
|>
|> * lor error *
|>
|> -- Executing Dial("SIP/1000-1de2", "Unicall/g1/40020022|40|Ttr") 
|> in new stack Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 
|> unicall_report: MFC/R2
|> UniCall/1 Call control(1)
|> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: 
|> MFC/R2
|> UniCall/1 Make call
|> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: 
|> MFC/R2
|> UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 
|> WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
|> UniCall/1 0001  ->  [1/   1/Idle  /Idle ]
|> -- Called g1/40020022
|> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
|> Unicall/1 event Dialing
|> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1  <-   [1/  40/Seize /Idle ]
|> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1 4 on  ->  [2/  40/Group I   /Idle ]
|> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1 R2 prot. err. [2/  40/Group I   /DNIS  
|   ] cause
|> 32769 - T1 timed out
|> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1 4 off ->  [1/   1/Idle  /Idle ]
|> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
|> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
|> Unicall/1 event Protocol failure
|> -- Unicall/1 protocol error. Cause 32769 Apr 20 19:14:02 
|> WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
|> UniCall/1 Channel echo cancel
|> Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: 
|> MFC/R2
|> UniCall/1 Channel gains
|> Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: 
|> MFC/R2
|> UniCall/1 Channel switching
|> -- Hungup 'UniCall/1-1'
|>   == Everyone is busy/congested at this time (1:0/0/1)
|>   == Auto fallthrough, channel 'SIP/1000-1de2' status is 
|'CHANUNAVAIL'
|> Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1  <- 1011  [1/   1/Idle  /Idle ]
|> Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 
|unicall_report: MFC/R2
|> UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
|>
|> Jefferson Carvalho
|>
|>
|>
|>
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|>
|
|
|--
|"Su nombre es GNU/Linux, no solamente Linux, mas info en 
|http://www.gnu.org";
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[William M Conlon]

Fair enough.

Though -- this is more than a cable, since it at least has a voltage  
divider, if not a voltage requlator, to drop the supply voltage.  So  
you have a rather subtle potential failure mode.


Just my opinion, but I prefer to conflate POE and IEEE 802.3af.  I  
wouldn't want to buy an "ethernet" device that used a CAT-5 cable but  
only received Morse Code.


On Apr 21, 2006, at 4:11 PM, Kevin P. Fleming wrote:


William M Conlon wrote:


3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE phone,
IMHO.  Caveat emptor.


It is if you buy the PoE injector cable instead of the self-powered  
one.

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RE: [Asterisk-Users] Error installing oh323

[T. Shaw]

Your linker is looking for the ldap libraries.
Do you have them installed?
 
Terrelle
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro 
Mejía EvertszSent: Friday, April 21, 2006 3:36 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Error installing oh323


I'm 
running:
OS: FreeBSD 
6.0
Asterisk: 
1.2.4
Installing OH323: 
0.7.3
 
I have this error when 
compiling
 
chan_oh323.c: In function 
`reload_config':
chan_oh323.c:4677: warning: 
implicit declaration of function `sscanf'
chan_oh323.c: At top 
level:
chan_oh323.c:3244: warning: 
'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o 
chan_oh323.so chan_oh323.o \ -L../wrapper -loh323wrap_s \ 
-L/usr/src/openh323_v1_17_1/lib -lh323_FreeBSD_x86_r_s \ 
-L/usr/src/pwlib_v1_9_0/lib -lpt_FreeBSD_x86_r_s \
-lstdc++ -lldap -lldap_r 
-llber -lpthread -lssl -lcrypto -lexpat
 
/usr/bin/ld: 
cannot find -lldap
 
gmake[1]: *** 
[chan_oh323.so] Error 1
gmake[1]: Leaving directory 
`/usr/src/asterisk-oh323-0.7.3/asterisk-driver'
gmake: *** [subdirs_build] 
Error 1
 
Any 
ideas??
 
Thanks in 
advance.
 
Alejandro.
 
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Kevin P. Fleming]

William M Conlon wrote:

> 3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE phone,
> IMHO.  Caveat emptor.

It is if you buy the PoE injector cable instead of the self-powered one.
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[William M Conlon]

Beware.  Although the Polycom 501s will sink power over ethernet  
(that is they are powered by a cable pair within a cable that  
resembles ethernet), they are NOT IEEE 802.3af POE devices!  They  
work on 12 volts (I think -- haven't measured it) instead of 48VDC.   
So don't expect to buy a POE source and expect the phone to receive  
power just by plugging in a patch cable.  They must be powered by the  
Polycom voltage sources.


Nevertheless, here's what works for me:

Netgear FS108 :: Polycom injector cable :: RJ45 coupler :: patch  
cable :: Polycom 501


Some notes:
1.  The Polycom injector cable should be plugged into a POE port on  
the switch (the Netgear FS108 switch has both powered and unpowered  
ports), or the Polycom injector will not source power.

2.  The Netgear FS108 is NOT sourcing power.
3.  The patch cable is a 50-foot CAT5.
3.  To beat a dead horse, the Polycom 501 itself, is NOT a POE phone,  
IMHO.  Caveat emptor.


Bill

On Apr 21, 2006, at 8:07 AM, <[EMAIL PROTECTED]>  
<[EMAIL PROTECTED]> wrote:



Hi listers,
I am looking for people who have used Power over Ethernet  
switches, primarily in conjunction with Polycom IP 501's.  I've  
been looking at the Linksys SRW224P, since I've had good luck with  
the SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all  
have an offering, all of which vary in price.  I would appreciate  
any input people have to offer.


Thanks,

James

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Bill

William M. Conlon, P.E., Ph.D.
To the Point
345 California Avenue Suite 2
Palo Alto, CA 94306
   vox:  650.327.2175 (direct)
   fax:  650.329.8335
mobile:  650.906.9929
e-mail:  mailto:[EMAIL PROTECTED]
   web:  http://www.tothept.com

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Re: [Asterisk-Users] confused about iax and voip providers termination

[Moises Silva]

The VOIP provider does not actually care if you are making your calls
from a simple SoftPhone, or a complete PBX. Im not going to explain
all the possible combinations of connecting, but i guess your
confusion comes because you still dont get to the part where some one
explains to you what "Native Transfer" means. IAX is a very nice
protocol working behind firewalls, so, unless you configure Asterisk
properly, asterisk is going to make the initial connection from your
softphone to the VoIP provider, and then will transfer the call
directly, so youre phone and the provider talk without intervention of
Asterisk. However thats not a good thing if you want to to some
billing (because Asterisk wont realize when the call ends), so in
iax.conf you can configure the phone with "notransfer=yes" (please
check the name of the parameter, im not sure) so Asterisk will stay in
the middle of the call.

Best Regards

On 4/21/06, T. Shaw <[EMAIL PROTECTED]> wrote:
> Hey guys,
> I'm actively trying to get the "big" picture on how all this works and
> relates to each other.
> I've gone through some basic examples from the book and from the sample
> files just fine.
> Now, I've setup an account with a VOIP provider which does IAX termination
> (exgn.net)
>
> After getting an account and following their steps, I can make calls out
> using my IAX (cubix) and Sip (Xlite) phones.
> However, I'm a bit confused on the purpose on how my box asterisk box is
> involved. I completely turned off my Asterisk box, and made a call out using
> either of my softphones and I was successful. So I gathered that the entire
> point of "iax termination" is solely for INBOUND calls TO ME (such if I have
> a DID). Otherwise I'm just using them as a proxy to forward my sip traffic
> to them directly from my desktop.
>
> I got confused because all references I have seen regarding "iax
> termination" and such involved editing your local asterisk box configs as
> well as the client, but really no clear mention that your config changes
> only apply to INBOUND calls, and not needed if you want to just make
> OUTBOUND
> Sip calls. I want to do BOTH eventually, but since I still have this
> learning curve, it was just another stumble for me.
>
> Do I have the correct picture now?
> Thanks!
>
> Terrelle Shaw
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Mike Garey]

We've got the Linksys SRW224P here in the office, powering 5 Polycom
IP-501 phones with no problems or issues.  Bought it from
tigerdirect.ca 
(http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=1815261&sku=L48-2460)
for about $450 after instant rebate (rebate no longer available it
seems).  It does 15 watts per port when using 12 ports, or 7.5 watts
per port when using all 24.  We've also got the PoE switch and
asterisk server running off of a 780 watt APC BX-1200N UPS ($200),
which gives us 30 minutes of run time after a power failure.

According to the Polycom knowledgebase
(http://eknowledge.polycom.com/SRVS/CGI-BIN/WEBCGI.EXE/,/?St=48,E=3886523,K=5531,Sxi=0,Case=obj(1240)),
it states the following regarding power requirements for the IP500
series phones:

"From the Cisco switch, the default inline power allocation per port
is 10.0 watts (0.24 amps @42V). The IP500CS will initially report to
the switch its power requirements and then use 4W.
The SoundPoint IP does support Phantom Power and does require the use
of the supplied Polycom cable."

So hopefully I should be fine on the Linksys switch with only 7.5
watts per port, if the Polycom's only use 4 watts (although I'm a bit
confused about the above statement, in regards to the "initial power
requirements" - not sure what will happen if these phones report a
higher power requirement than what my linksys switch will give them,
even though it seems they only require 4 watts).

Mike


On 4/21/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> The Edgewater Networks 2402 is affordable, supports IEEE 802.3af and will
> also power Cisco legacy phones.  Also does VLAN tagging and other L3 stuff.
> Here's a thumbnail
>
> * Automated setup when used in conjunction with EdgeMarc and EdgeView
> * 24 x 10/100 Mbps Autosensing Ethernet ports
> * 2 x 10/100/1000 Mbps Autosensing, Ethernet uplinks or high speed
> server ports with RJ-45 or SFP Fiber connectors
> * In-line Ethernet power (802.3af) with automatic power device detection
> * In-line power for many legacy (non-802.3af) IP handsets
> * Per port optional power sourcing
> * Full wire-speed forwarding
> * L2/L3/L4 traffic management
> * 802.1Q VLAN support
> * 802.1p based Quality of Service (QoS) with priority queuing
> * Link aggregation (802.3ad)
> * Complies with IEEE802.3, IEEE802.3u, IEEE802.3x and IEEE802.3af
> * Broadcast storm control
> * Management: integration with EdgeView NMS and support for SNMP, RMON,
> http, Telnet and DHCP
>
>
> Cory Andrews
> Executive Vice President
> ++
> VoIPSupply.com
> PBXSelect.com
> ++
> 454 Sonwil Drive
> Buffalo, NY 14225
> voice - 800.398.VoIP X3402
> fax - 716.630.1548
> e - [EMAIL PROTECTED]
> m - 716.907.4059
> aim - B2Cory
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kristian
> Kielhofner
> Sent: Friday, April 21, 2006 4:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Power over Ethernet (PoE) switch
> recommendations
>
> [EMAIL PROTECTED] wrote:
> > Hi listers, I am looking for people who have used Power over Ethernet
> > switches, primarily in conjunction with Polycom IP 501's.  I've been
> > looking at the Linksys SRW224P, since I've had good luck with the
> > SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all have
> > an offering, all of which vary in price.  I would appreciate any
> > input people have to offer.
> >
> > Thanks,
> >
> > James
> >
>
> First some comments on other posts -
>
> Stay away from the D-Link switches.  The admin interface is
> horrible,
> and most only support PoE on half the ports.  Plus, most of their stuff
> comes configured with a static IP on varying RFC1918 networks -
> 192.168.3.1, 192.168.10.1, etc, etc.  Not a good idea.  Plus, in the
> $400+ price range they really should have a serial port.
>
> I will never buy managed Netgear switches again.  I had two
> FSM7326Ps
> and the flash got corrupted several times, on both of them.  The web
> interface also had some serious reliability issues, sometimes causing
> the switch to reboot, which is of course totally unacceptable.
>
> I have been happy with Cisco (anything), HP 2650s and the Linksys
> SRW224P (or whatever it is) for managed PoE.  If you have the money (but
> not ridiculous, Cisco money) go with the HP.  Better warranty, better
> interface, better support than Linksys or D-Link.  Plus, I think they
> give you full 75W per port, as opposed to the SRW224p which can only do
> half wattage on all of the ports or full wattage on half the ports
> (something like that).  It's late in the day (and the week), so you'll
> have to cut me some slack!
>
> For the money, the Linksys can't be beat!
>
> --
> Kristian Kielhofner
> ___
> --Bandwidth and Colocation provided by Ea

[Asterisk-Users] Error installing oh323

[Alejandro Mejía Evertsz]

I'm running:

OS: FreeBSD 6.0

Asterisk: 1.2.4

Installing OH323:
0.7.3

 

I have this error
when compiling

 

chan_oh323.c: In
function `reload_config':

chan_oh323.c:4677:
warning: implicit declaration of function `sscanf'

chan_oh323.c: At
top level:

chan_oh323.c:3244:
warning: 'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o
chan_oh323.so chan_oh323.o \ -L../wrapper -loh323wrap_s \
-L/usr/src/openh323_v1_17_1/lib -lh323_FreeBSD_x86_r_s \
-L/usr/src/pwlib_v1_9_0/lib -lpt_FreeBSD_x86_r_s \

-lstdc++ -lldap
-lldap_r -llber -lpthread -lssl -lcrypto -lexpat

 

/usr/bin/ld: cannot find -lldap

 

gmake[1]: ***
[chan_oh323.so] Error 1

gmake[1]: Leaving
directory `/usr/src/asterisk-oh323-0.7.3/asterisk-driver'

gmake: ***
[subdirs_build] Error 1

 

Any ideas??

 

Thanks in
advance.

 

Alejandro.

 






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[Asterisk-Users] Very high size-32 usage

[Anthony Rodgers]

Hi there,

Has anyone noticed very high size-32 allocations in Asterisk servers 
with Digium hardware installed? Here is output from /proc/slabinfo:


size-32   23763586 23763586 32  1191 : tunables  120   
600 : slabdata 199694 199694  0


Here is the summary and first few rows from slabtop:

 Active / Total Objects (% used): 23850372 / 23890412 (99.8%)
 Active / Total Slabs (% used)  : 204139 / 204139 (100.0%)
 Active / Total Caches (% used) : 95 / 134 (70.9%)
 Active / Total Size (% used)   : 756630.62K / 760089.77K (99.5%)
 Minimum / Average / Maximum Object : 0.01K / 0.03K / 128.00K

  OBJS ACTIVE  USE OBJ SIZE  SLABS OBJ/SLAB CACHE SIZE NAME
23764300 23764241 -80%0.03K 199700  119798800K size-32
  5085   5085 100%0.68K   10175  4068K ext3_inode_cache
 51075  20557  40%0.05K681   75  2724K buffer_head
  8008   3666  45%0.27K572   14  2288K radix_tree_node
  9936   9863  99%0.16K432   23  1728K dentry_cache
  8463   8463 100%0.12K273   31  1092K size-128
   256256 100%3.00K1282  1024K biovec-(256)

As you can see, almost 800MB of memory on this box is taken up with 
size-32 pages.


This particular server is a single CPU box running Asterisk 1.2.5 and 
Zaptel 1.2.4 on RHEL4 and is a low-use, test box. Our two production 
boxes are dual 3.4GHz Xeons running Asterisk 1.2.1 and Zaptel 1.2.1 on 
RHEL4 SMP and exhibit the same issue (it was running into oom-killer 
problems with low LOWMEM on one of them that triggered all of this).


Interestingly, we have an identical server to our test server that does 
not have Asterisk or Zaptel installed, and it does not display this 
issue.


Has anyone else encountered this issue? What does your slabtop look 
like?


Any thoughts or ideas would be appreciated.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

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Re: [Asterisk-Users] extension match sip address

[Jon-o Addleman]

On Fri, Apr 21, 2006 at 02:39:44PM -0600, Alejandro Mejía Evertsz spake thusly:
> exten => _sipuserX.,1,blah
> to match sipuser01, sipuser99... ?
> or
> exten => sipuser01,1,blah
> to match sipuser01 only ?
> 
> Not to usefull when you want to match domain also :S
> Eg [EMAIL PROTECTED]

Looks like I'll just do something like this - I realized I can have the
PHP that's starting the call just add SIP to the beginning of the
extension, and then asterisk can just use a substring to chop off the
beginning of it.

Thanks for the suggestions!
-- 
Jon-o Addleman - http://redowl.dyndns.org
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[Asterisk-Users] test - ignore

[Rich Adamson]

lost our primary dns & email server, but now recovered and testing.

R.

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RE: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Cory Andrews]

The Edgewater Networks 2402 is affordable, supports IEEE 802.3af and will
also power Cisco legacy phones.  Also does VLAN tagging and other L3 stuff.
Here's a thumbnail

* Automated setup when used in conjunction with EdgeMarc and EdgeView
* 24 x 10/100 Mbps Autosensing Ethernet ports
* 2 x 10/100/1000 Mbps Autosensing, Ethernet uplinks or high speed
server ports with RJ-45 or SFP Fiber connectors
* In-line Ethernet power (802.3af) with automatic power device detection
* In-line power for many legacy (non-802.3af) IP handsets
* Per port optional power sourcing
* Full wire-speed forwarding
* L2/L3/L4 traffic management
* 802.1Q VLAN support
* 802.1p based Quality of Service (QoS) with priority queuing
* Link aggregation (802.3ad)
* Complies with IEEE802.3, IEEE802.3u, IEEE802.3x and IEEE802.3af
* Broadcast storm control
* Management: integration with EdgeView NMS and support for SNMP, RMON,
http, Telnet and DHCP


Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Friday, April 21, 2006 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Power over Ethernet (PoE) switch
recommendations

[EMAIL PROTECTED] wrote:
> Hi listers, I am looking for people who have used Power over Ethernet
> switches, primarily in conjunction with Polycom IP 501's.  I've been
> looking at the Linksys SRW224P, since I've had good luck with the
> SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all have
> an offering, all of which vary in price.  I would appreciate any
> input people have to offer.
> 
> Thanks,
> 
> James
> 

First some comments on other posts -

Stay away from the D-Link switches.  The admin interface is
horrible, 
and most only support PoE on half the ports.  Plus, most of their stuff 
comes configured with a static IP on varying RFC1918 networks - 
192.168.3.1, 192.168.10.1, etc, etc.  Not a good idea.  Plus, in the 
$400+ price range they really should have a serial port.

I will never buy managed Netgear switches again.  I had two
FSM7326Ps 
and the flash got corrupted several times, on both of them.  The web 
interface also had some serious reliability issues, sometimes causing 
the switch to reboot, which is of course totally unacceptable.

I have been happy with Cisco (anything), HP 2650s and the Linksys 
SRW224P (or whatever it is) for managed PoE.  If you have the money (but 
not ridiculous, Cisco money) go with the HP.  Better warranty, better 
interface, better support than Linksys or D-Link.  Plus, I think they 
give you full 75W per port, as opposed to the SRW224p which can only do 
half wattage on all of the ports or full wattage on half the ports 
(something like that).  It's late in the day (and the week), so you'll 
have to cut me some slack!

For the money, the Linksys can't be beat!

--
Kristian Kielhofner
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RE: [Asterisk-Users] extension match sip address

[Alejandro Mejía Evertsz]

Nop.

. matches one or more from the previous carácter

More info... 
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Eric
"ManxPower" Wieling
Enviado el: Friday, April 21, 2006 2:36 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] extension match sip address

Jon-o Addleman wrote:
> Is there a way to have an extension match on a sip address? I've tried
> the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is 
> no
> good.
> 
> Is there a better way? 

. stops a pattern match.


-- 
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Kristian Kielhofner]

[EMAIL PROTECTED] wrote:

Hi listers, I am looking for people who have used Power over Ethernet
switches, primarily in conjunction with Polycom IP 501's.  I've been
looking at the Linksys SRW224P, since I've had good luck with the
SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all have
an offering, all of which vary in price.  I would appreciate any
input people have to offer.

Thanks,

James



First some comments on other posts -

	Stay away from the D-Link switches.  The admin interface is horrible, 
and most only support PoE on half the ports.  Plus, most of their stuff 
comes configured with a static IP on varying RFC1918 networks - 
192.168.3.1, 192.168.10.1, etc, etc.  Not a good idea.  Plus, in the 
$400+ price range they really should have a serial port.


	I will never buy managed Netgear switches again.  I had two FSM7326Ps 
and the flash got corrupted several times, on both of them.  The web 
interface also had some serious reliability issues, sometimes causing 
the switch to reboot, which is of course totally unacceptable.


	I have been happy with Cisco (anything), HP 2650s and the Linksys 
SRW224P (or whatever it is) for managed PoE.  If you have the money (but 
not ridiculous, Cisco money) go with the HP.  Better warranty, better 
interface, better support than Linksys or D-Link.  Plus, I think they 
give you full 75W per port, as opposed to the SRW224p which can only do 
half wattage on all of the ports or full wattage on half the ports 
(something like that).  It's late in the day (and the week), so you'll 
have to cut me some slack!


For the money, the Linksys can't be beat!

--
Kristian Kielhofner
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RE: [Asterisk-Users] extension match sip address

[Alejandro Mejía Evertsz]

exten => _sipuserX.,1,blah
to match sipuser01, sipuser99... ?
or
exten => sipuser01,1,blah
to match sipuser01 only ?

Not to usefull when you want to match domain also :S
Eg [EMAIL PROTECTED]
 

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jon-o Addleman
Enviado el: Friday, April 21, 2006 2:25 PM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] extension match sip address

Is there a way to have an extension match on a sip address? I've tried
the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no
good.

Is there a better way? 
-- 
Jon-o Addleman - http://redowl.dyndns.org
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Re: [Asterisk-Users] extension match sip address

[Eric \"ManxPower\" Wieling]

Jon-o Addleman wrote:

Is there a way to have an extension match on a sip address? I've tried
the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no
good.

Is there a better way? 


. stops a pattern match.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [Asterisk-Users] Definitive list of sounds

[Eric \"ManxPower\" Wieling]

Kristian Kielhofner wrote:

Steve Kennedy wrote:

Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.

There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.

Any help appreciated.


Steve



Steve,

The format is a little funky, but it should work:

http://mirror.astlinux.org/sounds/en-prompts.txt


Or you can look at /path/to/src/asterisk/sounds.txt  A similar file 
exists for asterisk-addons when you check it out from SVN.

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Re: [Asterisk-Users] Definitive list of sounds

[Kristian Kielhofner]

Steve Kennedy wrote:

Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.

There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.

Any help appreciated.


Steve



Steve,

The format is a little funky, but it should work:

http://mirror.astlinux.org/sounds/en-prompts.txt

--
Kristian Kielhofner
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[Asterisk-Users] confused about iax and voip providers termination

[T. Shaw]

Hey guys, 
I'm actively trying to get the "big" picture on how all this works and
relates to each other.
I've gone through some basic examples from the book and from the sample
files just fine.
Now, I've setup an account with a VOIP provider which does IAX termination
(exgn.net)

After getting an account and following their steps, I can make calls out
using my IAX (cubix) and Sip (Xlite) phones.
However, I'm a bit confused on the purpose on how my box asterisk box is
involved. I completely turned off my Asterisk box, and made a call out using
either of my softphones and I was successful. So I gathered that the entire
point of "iax termination" is solely for INBOUND calls TO ME (such if I have
a DID). Otherwise I'm just using them as a proxy to forward my sip traffic
to them directly from my desktop.

I got confused because all references I have seen regarding "iax
termination" and such involved editing your local asterisk box configs as
well as the client, but really no clear mention that your config changes
only apply to INBOUND calls, and not needed if you want to just make
OUTBOUND 
Sip calls. I want to do BOTH eventually, but since I still have this
learning curve, it was just another stumble for me.

Do I have the correct picture now?
Thanks!

Terrelle Shaw
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[Asterisk-Users] extension match sip address

[Jon-o Addleman]

Is there a way to have an extension match on a sip address? I've tried
the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no
good.

Is there a better way? 
-- 
Jon-o Addleman - http://redowl.dyndns.org
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[Asterisk-Users] server choice

[issam]

hello
I will buy a server to make an IVR solution with 
asterisk and a te110p T1/E1 digium card.
I have two options:
1/ HP Proliant ML370 G4 : Xeon 64bits 3,2Ghz, 1Go 
Ram, 3 disks SCSI 73Go
2/ Dell PowerEdge 2800: Xeon 64bits 3Ghz, 1Go Ram, 
3 disks SCSI 73Go
 
I use linux fedora core 3 and I want a help to 
choose a good server to use with asterisk
 
Thank You 
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Re: [Asterisk-Users] Asterisk on Red Hat AS 4?

[Anthony Rodgers]

Hi Domenico,

We're using RHEL 4 ES with no obvious issues

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Apr 21, 2006, at 3:59 AM, Mimmus wrote:


Hi,
I'm planning to install a new Asterisk server with a Digium TE410P 
card.

Can I use Red Hat Advanced Server 4 (latest update)?
Is this a good choice?
Is recompiling Asterisk simple with kernel 2.6?

Thanks
--
Domenico Viggiani

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Re: [Asterisk-Users] still some moh troubles

[Anthony Rodgers]

Hi Bart,

If it's anything like the problem we had, you are probably getting what 
sounds like screeching noises during MOH playback? We had this problem 
and made it go away by turning off hyperthreading in the server BIOS 
and starting Linux with noht - this was on a dual Xeon machine.


Hope this helps.

Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Apr 20, 2006, at 6:37 AM, Bart van Daal wrote:


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug 
Lytle

Sent: donderdag 20 april 2006 14:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] still some moh troubles

>>Bart van Daal wrote:
>> Hi,
>>
>> After following the suggestions on the mailing lists and the wiki 
I'm
>> still experiencing choppy moh. The song plays but with frequent 
noise

>> parts.
>>
>> - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the
>> test server.
>> - native moh with .gsm and .pcm formats (according to
>>  
>
>Actually, you'll want to use ulaw for Native MOH.
>
>CUT
>
>
>#!/bin/sh
>
>for filename in *mp3
>
>do
>
>eval filename=`echo $filename | cut -f1 -d.`
>
>echo Converting $filename
>
>sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample 
-ql

>
>done
>
>CUT
>
>Doug

Thanks for you suggestion Doug,
I've converted the files using your script to ulaw but experience the 
same

problem.
A thing I forgot to mention is that it only happens on calls passing 
the

trunks to the
cisco-routers that terminate to pstn so not on internal sip-sip calls.
Normal voice communication runs smoothly over the trunks it's only the 
moh

that causes some problems.

again, any pointers like those of Doug are very much appreciated

thanks!
Bart













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RE: [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering

[Steve Jones]

I had a similar problem with a GS101, although with mine, I could make
OUTBOUND calls from the phone, but because it wasn't registered, it
wouldn't ring if called.  I don't know the exact solution, but two
things I did was to tell it NOT to subscribe to MWI in the GS config
itself, and second, I upgraded the firmware.  I actually think the
solution was the MWI though. 

I didn't use the standard TFTP config for the upgrade, but I Don't have
the IP addresses I used handy - If you still can't get it to work, let
me know, and when I'm home, I'll look up the IP address.

-Steve

-Original Message-
From: Marcel Hecko [mailto:[EMAIL PROTECTED] 
Sent: Friday, April 21, 2006 2:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering

Hello,
I have a problem with one of three [topic] phones. The phone, which is
on the LAN in the same subnet as Asterisk, keeps unregistering from the
Asterisk server. Whan it is unregistered there is no way to make a phone
call from it, but once it is rang by any other of the phones it
registers to Asterisk again. The other two are absolutely fine.

The problematic one [ecco] puts this messages into messages log file:

Apr 21 15:27:23 NOTICE[1099] chan_sip.c: Auto-congesting SIP/ecco-9091
Apr 21 15:27:23 WARNING[1077] channel.c: Avoided initial deadlock for
'0x8129f38', 10 retries!

sip.conf:
[ecco]
type=friend
username=ecco
defaultip=10.10.129.31
host=dynamic
nat=no
canreinvite=no
qualify=yes
dtmfmode=rfc2833


Can somebody please explain what these messages mean?
What is '0x8129f38'?
Can somebody please post working sip.conf (full) for Grandstream (101)
phones. I have discussed this topic with Google for quite a long time,
but with no results.

Thank you very much.
Marcel

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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Olivier Krief wrote:
> To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best
> software combination ? Asterisk and Hylafax ?

You can use any combination of CAPI based software in parallel. You just 
need to create the rules for which application shall act on which 
number/service.
 
> Shall we then allocate destination numbers and or ports for each of those 2
> applications ?

That depends on what you want to do. On incoming calls, each application 
gets the information about destination number and bearer-capability 
(service). If you can separate your services with these information, then 
all is fine. Just configure the applications not to act on the others 
numbers/services.
For Fax and Voice on the same number it is not possible, because the 
bearer-capability may be the same. So one application to handle both
is needed.
This can be done with Asterisk/chan-capi. Receiving fax (via CAPI without 
spandsp) is a feature supported chan-capi if the isdn driver supports it.
 
> And if you want to offer to every user, a unique extension for fax and
> voice, would it still be possible to forward calls from voice application to
> fax application (for outgoing faxes, the fax application can use its own
> ressources) ?

Outgoing is not a problem, just make sure that one application won't use too 
many channels and block other appliactions...

Having a unique fax extension for each user can be done with 
Asterisk/chan-capi allone and with another fax-software like Hylafax as 
well.
But if you want to forward a call (which was already accepted by Asterisk) 
to another CAPI application, it is not possible. (Well, Eicon has a special 
driver which can do a lot of CAPI extensions, but I did not try this yet).
So if you want to do that, I suggest using just chan-capi for receiving 
faxes and maybe another application for sending faxes.

Armin
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Matt Roth]

We're using Cisco Catalyst 3560 Series 48 port PoE switches.  So far, 
*they just work*.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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RE: [Asterisk-Users] Codec problem from SIP to H323

[Alejandro Mejía Evertsz]

I tried by just upgrading to Ast1.2.4 but same problem.
Then I tried to install OH323 but I have this error when compiling :S

chan_oh323.c: In function `reload_config':
chan_oh323.c:4677: warning: implicit declaration of function `sscanf'
chan_oh323.c: At top level:
chan_oh323.c:3244: warning: 'update_call_ids' defined but not used
gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o \
-L../wrapper -loh323wrap_s \
-L/usr/src/openh323_v1_17_1/lib -lh323_FreeBSD_x86_r_s \
-L/usr/src/pwlib_v1_9_0/lib -lpt_FreeBSD_x86_r_s \
-lstdc++ -lldap -lldap_r -llber -lpthread -lssl -lcrypto -lexpat
/usr/bin/ld: cannot find -lldap
gmake[1]: *** [chan_oh323.so] Error 1
gmake[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.3/asterisk-driver'
gmake: *** [subdirs_build] Error 1

Does anyone know how to get rid of that?

I'm running:
OS: FreeBSD 6.0
Asterisk: 1.2.4
OH323: 0.7.3

Thanks in advance.

Alejandro.

-Mensaje original-
De: Oliver Vermeulen [mailto:[EMAIL PROTECTED] 
Enviado el: Wednesday, April 19, 2006 4:09 PM
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
CC: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] Codec problem from SIP to H323

Try to upgrade asterisk to version 1.2.4 

Are you using OH323 or H323 ?

I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323
and everything worked fine.

Cheers,
Oliver



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Mejía Evertsz
Sent: Thursday, April 20, 2006 12:44 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Codec problem from SIP to H323

Hello.

I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:

- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf has "disallow=all & allow=g729"
- h323.conf has "disallow=all & allow=g729"

The problem:

[SIPphone]  [sip.conf]  [h323.conf]
[H323gw]
g729--->allow=g729  --->allow=g729  --->g729

When I dial to the gateway from the SIPphone using g729 as my sip phone's
default codec I get:

-- Executing Dial("SIP/amejia-8be1", "H323/[EMAIL PROTECTED]") in new stack
Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator
path exists for channel type H323 (native 4) to 256
Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to
create channel of type 'H323' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)


I don't get it why is it trying to "translate" anything. There's nothing to
translate, cause I'm using g729 in both ends.
Well, to make it more interesting, I tried this way:

[SIPphone]  [sip.conf]  [h323.conf]
[H323gw]
g711--->allow=all   --->allow=all   --->g729

This way, it passes the call to the gateway just giving a waring that it
can't find a codec to translate. But at least it passes the call.
It rings on the other side, and of course as I don't have any g729 licenses
installed it drops the call when answered.

-- Executing Dial("SIP/amejia-1fc8", "H323/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a
codec translation path from g729 to ulaw
-- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8
-- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8
-- H323/H323gw-2 is ringing
-- H323/H323gw-2 answered SIP/amejia-1fc8
Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible:
No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256)
Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop
call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2
  == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8'


Does anybody know how can I get rid of the problem I get on the first
scenario?
Why does it try to use codec 4 (g711u) if both ends are configured with
g729?

Please give me some light. I don't know what else to try.

Thank you all.

Alejandro Mejia

___

[Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering

[Marcel Hecko]

Hello,
I have a problem with one of three [topic] phones. The phone, which is
on the LAN in the same subnet as Asterisk, keeps unregistering from the
Asterisk server. Whan it is unregistered there is no way to make a phone
call from it, but once it is rang by any other of the phones it
registers to Asterisk again. The other two are absolutely fine.

The problematic one [ecco] puts this messages into messages log file:

Apr 21 15:27:23 NOTICE[1099] chan_sip.c: Auto-congesting SIP/ecco-9091
Apr 21 15:27:23 WARNING[1077] channel.c: Avoided initial deadlock for
'0x8129f38', 10 retries!

sip.conf:
[ecco]
type=friend
username=ecco
defaultip=10.10.129.31
host=dynamic
nat=no
canreinvite=no
qualify=yes
dtmfmode=rfc2833


Can somebody please explain what these messages mean?
What is '0x8129f38'?
Can somebody please post working sip.conf (full) for Grandstream (101)
phones. I have discussed this topic with Google for quite a long time,
but with no results.

Thank you very much.
Marcel
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Andrew Latham]

My favorite one is this one.
http://www.provantage.com/d-link-systems-des-1526~7DLNS046.htm



On 4/21/06, Steve Kennedy <[EMAIL PROTECTED]> wrote:
> On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:
>
> > D-link has a nice one, optional 5 year warranty on some of the
> > commercial stuff
>
> Though beware, some of the D-Link ones only have half the ports with
> PoE.
>
>
> Steve
>
>
> --
> NetTek Ltd  UK mob +44-(0)7775 755503
> UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
> Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
> Euro Tech News Blog http://eurotechnews.blogspot.com
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?

[Moises Silva]

that feature does not exists AFAIK, but you can request it in
bugs.digium.com, or offer some money to someone to include it for you.

Regards

On 4/21/06, Pimjai Wesnarat <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> I'm trying to add this function to my find-me application: when all
> available numbers are dialed in parallel , if any number is busy, take
> it at busy and go to voice mail.  I read the Dial() Application but
> there's nothing written about this. My question is, is it possible to do
> this with Asterisk?
>
> Thank you,
>
> Pim
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--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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Re: [Asterisk-Users] Jingle support - can we test the feature ?

[Robert Rozman]

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


Sent: Friday, April 21, 2006 11:43 AM
Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ?




On 20 Apr 2006, at 16:39, Robert Rozman wrote:



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


Sent: Thursday, April 20, 2006 4:18 PM
Subject: Re: [Asterisk-Users] Jingle support - can we test the  feature ?


we would like to build IM-Voice community for our students around 
Asterisk,

Jingle, Jabber.

Can we already test those features ?  Anyone already running such 
setup? Any

more info ?

Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/

There is an Asterisk-plugin that update your status automagically when
you're on the phone
--


Hi,

thanks for pointer. I know for that project, but reading about  Jingle, 
Jabber and Asterisk integration it seems not so interesting  for me at 
the moment...


Regards,





So what aspect of Jingle, Jabber and Asterisk  did you mean in your 
original post ?


Well I've read few general interviews and articles about integration of 
Jingle protocol and Asterisk. There is also IAX version of specification for 
audio transport. There is asterisk-xmpp effort. The main thing at least in 
my opinion would be that I could have network of Asterisk servers, and user 
could use integrated client that would give presence, IM and audio 
communication in Asterisk compatible way..


So I'm curious if anyone has made and tests or has more info on that 

Regards,

Rob. 


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Re: [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.

[Jeremy Parr]

On 4/21/06, Rick Smith <[EMAIL PROTECTED]> wrote:
>
> This is coming from an * noob. :)
>
> I've got two customers, they both are replacing their phone systems with
> VOIP, and we need to retain both their existing dialplans.
>
> One has 5 extensions starting at 100, and the other has 10 extensions,
> starting at 100.
>
> Is there a way to have the same extension number twice in the same
> asterisk system ?
>
> They will have different incoming DIDs of course.
>
> I don't want them to be able to see / hear / feel / dial each other
> internally, either.  They must remain completely independent.
>
> If anyone's got pointers in a Wiki or PDF somewhere, let me know.

Dead easy. Just put them in different contexts.
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[Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.

[Rick Smith]

This is coming from an * noob. :)

I've got two customers, they both are replacing their phone systems with 
VOIP, and we need to retain both their existing dialplans.


One has 5 extensions starting at 100, and the other has 10 extensions, 
starting at 100.


Is there a way to have the same extension number twice in the same 
asterisk system ?


They will have different incoming DIDs of course.

I don't want them to be able to see / hear / feel / dial each other 
internally, either.  They must remain completely independent.


If anyone's got pointers in a Wiki or PDF somewhere, let me know.

Thanks
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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Jens Vagelpohl]

On 21 Apr 2006, at 18:21, Olivier Krief wrote:

To benefit from DIVA Server 4BRI fax hardware capabilities, what is  
the best software combination ? Asterisk and Hylafax ?


Shall we then allocate destination numbers and or ports for each of  
those 2 applications ?


And if you want to offer to every user, a unique extension for fax  
and voice, would it still be possible to forward calls from voice  
application to fax application (for outgoing faxes, the fax  
application can use its own ressources) ?


I run the combination of Asterisk and Hylafax, and it is easy for me  
because I have 3 incoming MSNs. Both Asterisk and Hylafax will "see"  
all calls, but Asterisk only has two of the three MSNs configured.  
The fax number is ignored by Asterisk, so Hylafax answers that after  
a couple rings.


This works perfectly fine, but unfortunately I fell into the "DIVA V- 
BRI doesn't do FAX" trap, I bought a V-BRI first. It's now sitting on  
the shelf, unused. The Asterisk server is using the standard BRI  
card, which I bought off eBay.


IMHO the Eicon website should carry more prominent warnings/ 
explanations about the lack of FAX capabilities for the V-BRI cards.


jens

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[Asterisk-Users] Easier install of QueueMetrics on [EMAIL PROTECTED]

[Lenz]

Hello list,
we are testing an easier way to install QueueMetrics on an [EMAIL PROTECTED]  
box (or any other CentOS/RHEL) using the yum package manager.

This is still experimental, so it may as well work as not work.
We are looking for testers who are willing to try this at home  and any  
feedback would be helpful.


This should be it:

wget -P /etc/yum.repos.d http://yum.loway.it/loway.repo

yum install queuemetrics

or a longer version is here: http://astrecipes.net/index.php?n=182

QueueMetrics is a full-blown call center monitoring and reporting system,  
and is available for free to smaller CCs, home users and individual  
enthusiasts.

Any comment is welcome!
l.

--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Geoff Manning]

On 4/21/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hi listers,I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's.  I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price.  I would appreciate any input people have to offer.
We have been using a 3COM 2226 PWR Plus and have had no issues 
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Re: [Asterisk-Users] MWI in multi-PBX setup

[C F]

It really depends on the PBX in use. Avaya uses DTMF tones to light
the MWI, you can find examples on the wiki on how to do it. In which
case you shouldn't have any problem doing it.
Most of the bigger phone systems I have worked with allow the same
thru simple DTMF tones.

On 4/21/06, Olivier Krief <[EMAIL PROTECTED]> wrote:
> Has anyone tried to set Message Waiting Indicators up when public network
> access and voicemail service are managed by an Asterisk server TDM-connected
> to a legacy PBX serving analog and digital phones ?
>
> For instance:
>
> Location 1:
> - 200 users on a legacy PBX
> - among those users, 50 have access to voicemail service
> - TDM trunk to Location 2
>
> Location 2:
> - 100 users on Asterisk
> - PSTN access
> - TDM trunk to Location 1
>
> Does it make any sense to service all 300 users with MWI from the * server ?
> How does it work ?
> How could you light legacy PBX phones from Astrisk voicemail ?
>
> Regards
>
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>
>
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[Asterisk-Users] wellgate FXO unit

[Jerry Geis]

Anyone know how to set the wellgate unit so incoming calls
pass on directly to asterisk?

Right now incoming calls ring twice and I hear a recording saying enter
the extension. If I go enter the extension it goes on to asterisk just fine.

I just want the incoming call to go directly onto asterisk.

Anyone found that out?

Jerry
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[Asterisk-Users] MWI in multi-PBX setup

[Olivier Krief]

Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ?For instance:
Location 1:- 200 users on a legacy PBX- among those users, 50 have access to voicemail service- TDM trunk to Location 2Location 2:- 100 users on Asterisk- PSTN access- TDM trunk to Location 1
Does it make any sense to service all 300 users with MWI from the * server ?How does it work ?How could you light legacy PBX phones from Astrisk voicemail ?Regards 
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RE: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Chad Osmond]

SMC 6824MPE.. Does 24 ports POE, with 2x 1GB uplinks, (RJ45 or  GBIC)
L3 Managed switch.

We've got four of them here, and I think they're great, the cost was
really reasonable.

Ingram no longer lists the MPE model, but it should be available still.

Chad

-Original Message-
Hi listers,
I am looking for people who have used Power over Ethernet switches,
primarily in conjunction with Polycom IP 501's.  I've been looking at
the Linksys SRW224P, since I've had good luck with the SRW224 in our
office.  However, Nortel, Cisco, Adtran, etc. all have an offering, all
of which vary in price.  I would appreciate any input people have to
offer.
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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Olivier Krief]

To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ?Shall we then allocate destination numbers and or ports for each of those 2 applications ?
And if you want to offer to every user, a unique extension for fax and voice, would it still be possible to forward calls from voice application to fax application (for outgoing faxes, the fax application can use its own ressources) ?
Cheers
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Re: [Asterisk-Users] Call recording

[Jonathan Addleman]

Wai Wu wrote:
> I notice those options. However, I was looking to start the recording
> through a third party control program. I know I can do this via
> chanspy, but is there better way?

Not that I know of... I was looking for something kind of similar, and
ended up actually using a conference, and adding an ICES streaming
channel to it when I want to start recording. Since I'm actually trying
to use network streams, this is great for me, but might not be so good
for you...

-- 
Jon-o Addleman

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Re: [Asterisk-Users] channels change names

[Jonathan Addleman]

Peter Fern wrote:
> Probably because the Local proxy channel drops out once the two sides 
> have been bridged.  If you want the Local chan to stay up, use the /n 
> parameter and the local channel won't perform the native transfer.  This 
> does have it's own problems, but should do what you want.

Thanks for the tip! Seems to work fine for now. I'll look into something
with a variable for the future, but since I'm just getting the
information directly from the 'meetme list' command, it would take a bit
of rewriting to do that. For now, the quick-and-dirty works perfectly,
it seems.

-- 
Jon-o Addleman

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Re: [Asterisk-Users] Polycom MWI

[Eric \"ManxPower\" Wieling]

how about, in sip.conf, [EMAIL PROTECTED] in 
the [section] for that device?




Bill Gibbs wrote:

Put your voicemailbox number (usually extension) in the 1.subscribe field.
 
Bill




From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 4/20/2006 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Polycom MWI


I have tried everything from voip-info and I still cant get the Polycom 501/601 
to display the MWI indicator light. Everything else works just fine. I am using 
FreePBX set to users and devices mode. Here is the MWI section of the 
phonexxx.cfg file:
 
 
msg.mwi.1.subscribe="" 
msg.mwi.1.callBackMode="contact" 
msg.mwi.1.callBack="*97" 
msg.mwi.2.subscribe="" 
msg.mwi.2.callBackMode="disabled" 
msg.mwi.2.callBack="" 
msg.mwi.3.subscribe="" 
msg.mwi.3.callBackMode="disabled" 
msg.mwi.3.callBack="" 
msg.mwi.4.subscribe="" 
msg.mwi.4.callBackMode="disabled" 
msg.mwi.4.callBack="" 
msg.mwi.5.subscribe="" 
msg.mwi.5.callBackMode="disabled" 
msg.mwi.5.callBack="" 
msg.mwi.6.subscribe="" 
msg.mwi.6.callBackMode="disabled" 
msg.mwi.6.callBack=""/> 
 
   
 
i have also tried
 
msg.mwi.1.callBackMode="register"
 
 
Kerry Garrison

Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]  
http://www.techdatapros.com   
 





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[Asterisk-Users] Asterisk FAX-to-Email

[Wasif]

Hi,

How can we change the FROM address when Asterisk sends mail (in FAX-to-Email
feature). For example it is sending [EMAIL PROTECTED] in FROM
address; I need to change it to [EMAIL PROTECTED] 

Any help?



Wazb

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Re: [Asterisk-Users] problem with TE205

[Remco Barende]

Weird, I just received a new TE210P card (should be identical only 3.3v) 
but I cannot find any info on jumper settings om the Digium site?


But then again the installation info on the Digium site really sucks.


On Fri, 21 Apr 2006, Rob Lith wrote:


Jumpers must still be on for E1 mode.
Rob

On 21/04/06, Remco Barende <[EMAIL PROTECTED]> wrote:



Hello,

I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have

my

jumper set (i.e closed to use the E1 facility.)


Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were
completely obsolete now?

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RE: [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application??

[kevin ling]

Hi,

Check the script. You can assign the voice by -n option, e.g.,

/opt/swift/bin/swift -n Diane 

Kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane Young
Sent: Friday, April 21, 2006 9:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to select Ceptral's Voice in
Asterisk'sSwift application??

Quoting Pimjai Wesnarat <[EMAIL PROTECTED]>:

> Hi,
>
> I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
> It works fine when I have just 1 voice installed. Now I have 2 voices 
> in the same language installed but I can't seem to find the way to 
> select which voice to use in Swift's application in Asterisk. Does anyone
know??

[cepstral-demo]
exten => s,1,Answer
exten => s,n,wait(1)
exten => s,n,Cepstral(Hello and welcome to the world
of text to speech using Cepstral.  My name is Duchess.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Walter.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Shouty.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is William.) exten =>
s,n,Cepstral(Hello and welcome to the world of text
to speech using Cepstral.  My name is Whispery.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Robin.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Linda.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Emily.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Diane.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is David.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Duncan.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Damien.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Callie.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Dog.) exten =>
s,n,Cepstral(Hello and welcome to the world of text to
speech using Cepstral.  My name is Amy.)


This message was sent using IMP, the Internet Messaging Program.
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[Asterisk-Users] Asterisk FAx-to-Email

[Wasif]

-Original Message-
From: Wasif [mailto:[EMAIL PROTECTED] 
Sent: Thursday, April 20, 2006 4:25 PM
To: 'asterisk-users@lists.digium.com'
Subject: Asterisk FAx-to-Email


Hi,


I get error when my DID hit to asterisk box which I am using for FAX to
Email Service. Sometimes Fax goes through but mostly I get communication
error on Fax Machine and on Asterisk I get Comfort noise support incomplete
in Asterisk (RFC 3389) error.

I am using SIP with G711. My Did provider cannot turn off VAD and Echo from
his side, so is there any option or setting I can do at my side to make FAX
service more reliable



Thanks

Wazb

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Re: [Asterisk-Users] roundrobin strategy in queues not working as described?

[Josué Conti]

Hi Jim,
The function roundrobin makes with that asterisk directs the calls for the next free agent, but not orderly. I use the same strategy and functions here very well. The difference is that only use the functions of agent loginok and agent loginoff.

I wait to have helped
Good luck
Regads
Josué 
2006/4/21, Jim Rice <[EMAIL PROTECTED]>:
I have set up an operator queue for our receptionist.That way, if she takes a break or is out, by logging out of the queue,
calls to the "Operator" can be handled by other agents.I have set strategy = roundrobin in queues.conf.According to "the book" ATFoT, roundrobin always starts with the firstagent in the queue.  This is the desired result.  I want all calls to
start there, and if she is busy or does not answer, calls should go tothe next agent logged into the queue.Yet, I am seeing it behave as if it were "rrmemory".I called the Operator while she was not busy, and it rang the other
agent.  I called again, and this time it came to her.*CLI> show queue operatoroperator has 0 calls (max unlimited) in 'roundrobin' strategy (1sholdtime), W:0, C:1, A:4, SL:0.0% within 0s  Members:
 Agent/200 (Not in use) has taken 1 calls (last was 173 secs ago) Agent/110 (Unavailable) has taken no calls yet Agent/246 (Not in use) has taken 1 calls (last was 302 secs ago) Agent/140 (Unavailable) has taken no calls yet
  No CallersAny ideas?--Jim Riceby Design Publishing11626 N. Tracey RoadHayden, Idaho  83835___--Bandwidth and Colocation provided by 
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[Asterisk-Users] SIP domain in Asterisk

[Joao Pereira]

Hello to all
Can someone tell me if its possible to implement a SIP domain with 
Asterisk (im trying with [EMAIL PROTECTED]).

With a SIP domain I mean:
-users having URIs with [EMAIL PROTECTED] ( instead of [EMAIL PROTECTED] )
-being able to reach our users anywhere in the world with SIP URIs (and 
the help of SRV records)
-the possibility of dialing [EMAIL PROTECTED] and route the calls 
through the Internet


Can this be done?

Thanks
Joao Pereira
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[Asterisk-Users] roundrobin strategy in queues not working as described?

[Jim Rice]

I have set up an operator queue for our receptionist.
That way, if she takes a break or is out, by logging out of the queue,
calls to the "Operator" can be handled by other agents.

I have set strategy = roundrobin in queues.conf.

According to "the book" ATFoT, roundrobin always starts with the first
agent in the queue.  This is the desired result.  I want all calls to
start there, and if she is busy or does not answer, calls should go to
the next agent logged into the queue.

Yet, I am seeing it behave as if it were "rrmemory".

I called the Operator while she was not busy, and it rang the other
agent.  I called again, and this time it came to her.

*CLI> show queue operator
operator has 0 calls (max unlimited) in 'roundrobin' strategy (1s
holdtime), W:0, C:1, A:4, SL:0.0% within 0s
   Members:
  Agent/200 (Not in use) has taken 1 calls (last was 173 secs ago)
  Agent/110 (Unavailable) has taken no calls yet
  Agent/246 (Not in use) has taken 1 calls (last was 302 secs ago)
  Agent/140 (Unavailable) has taken no calls yet
   No Callers

Any ideas?

-- 
Jim Rice
by Design Publishing
11626 N. Tracey Road
Hayden, Idaho  83835

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[Asterisk-Users] HANGUPCAUSE on SIP channels

[Eric Futch]

Hopefully I'm not just missing some little detail here.  We're trying to 
set the HANGUPCAUSE on SIP channels to have our softswitch play the proper 
recording instead of answering the call on Asterisk to play the message. 
It appears that no matter what the HANGUPCAUSE is set to, Asterisk always 
just sends "603 Declined".


I looked through the source code briefly and it appears that it *should* 
work.  It would be helpful to know if anyone actually uses this feature 
and if it is working properly for them before we go through with fully 
debugging and patching this to work for us.


Here is the our test extension from extensions.conf:
exten => 9218,1,Set(HANGUPCAUSE=1)
exten => 9218,2,Hangup

According to hangup_cause2sip in chan_sip.c a HANGUPCAUSE of 1 should 
cause Asterisk to reply to the softswitch with a "404 Not Found" SIP 
message.  That doesn't seem to be the case, however.


Here is a bit of the verbose console output:
(Please note that I added some extra ast_log calls to the source code to 
generate some extra debugging information.)


Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: 
chan=SIP/nyct-901-539f, name=SIPURI, value=sip:[EMAIL PROTECTED]:5060
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: 
chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: 
chan=SIP/nyct-901-539f, name=SIPUSERAGENT, 
value=PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041
Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: 
chan=SIP/nyct-901-539f, name=SIPCALLID, 
[EMAIL PROTECTED]

-- Executing Set("SIP/nyct-901-539f", "HANGUPCAUSE=1") in new stack
Apr 21 12:35:18 WARNING[16815]: pbx.c:5983 pbx_builtin_setvar_helper: 
chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1
Apr 21 12:35:18 WARNING[16815]: pbx.c:6057 pbx_builtin_setvar: 
chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1

-- Executing Hangup("SIP/nyct-901-539f", "") in new stack
Apr 21 12:35:18 WARNING[16815]: pbx.c:5548 pbx_builtin_hangup: 
chan->hangupcause=(null)
  == Spawn extension (nyct, 9218, 2) exited non-zero on 
'SIP/nyct-901-539f'
Apr 21 12:35:18 WARNING[16815]: chan_sip.c:2471 sip_hangup: 
ast->hangupcause=16 res=(null)


This is all on Asterisk 1.2.7.1.  Your line numbers may vary since there 
were some ast_log lines added.  Hopefully this makes some sense to 
someone.


Thanks for any help or input.

--
New York Connect Technical Support Staff
Eric Futch <[EMAIL PROTECTED]> (212) 293-2620
Weather for KNYC: Apr 21 11:51a EDT, 59F (15C), Fair, Humidity 49%
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[Asterisk-Users] Definitive list of sounds

[Steve Kennedy]

Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.

There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.

Any help appreciated.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [Asterisk-Users] problem with TE205

[Rob Lith]

Jumpers must still be on for E1 mode.RobOn 21/04/06, Remco Barende <[EMAIL PROTECTED]> wrote:
> Hello,>> I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
> jumper set (i.e closed to use the E1 facility.)Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers werecompletely obsolete now?___
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Re: [Asterisk-Users] problem with TE205

[Remco Barende]

Hello,

I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my 
jumper set (i.e closed to use the E1 facility.)


Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were 
completely obsolete now?


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[Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?

[Pimjai Wesnarat]

Hi All,

I'm trying to add this function to my find-me application: when all 
available numbers are dialed in parallel , if any number is busy, take 
it at busy and go to voice mail.  I read the Dial() Application but 
there's nothing written about this. My question is, is it possible to do 
this with Asterisk?


Thank you,

Pim
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[Asterisk-Users] MoH issue

[Kevin Smith]

Hey everyone,

Hopefully I can describe the problem well enough so bear with me.

There are 3 companies that are tied into our asterisk server. Company A 
(us) uses the default settings for music on hold. Companies B and C 
however, want something different. For them I have when a call comes 
into their dial plan it sets the music on hold to their music and that 
seems to work. However, here is the problem. Calling out, it still plays 
the old on hold music.


Here is the situation, the 3 companies if they call each other us SIP 
and don't even touch the PRI, only outgoing calls outside the companies 
will do that. So I also would like if B called C, C's music on hold 
would be the one heard.


Here is how I started the dialplan.
[Empire-Outbound]
exten => _.,1,Answer()
exten => _.,n,SetMusicOnHold(OrigMusic)
exten => _.,n,Wait(2)
exten => _.,n,Goto(Empire-Outbound2,${EXTEN},1)

[Empire-Outbound2]
include => A-DirectDial ;Direct Dial Context
include => Empire-VoiceMail ;Voicemail context
include => Empire-Wildcard  ;Basic calling function


Does switch between contexts reset the moh? Or can I not change the moh 
for SIP channels and only on Zap?


Thanks,
Kevin
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Re: [Asterisk-Users] problem with TE205

[Infobox Peru]

Why do you comment these lines:

;channel=31-45
;channel=47-61

and put those in the zapata.conf?

bchan=31-45,47-60
dchan=46

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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Adam Lewis]

Netgear makes a 24 port Layer 3 Managed Switch, with PoE on all 24 ports.  It supports both IEEE 802.3af PoE as well as the proprietary Cisco PoE scheme (although the support for Cisco PoE is undocumented).  Got one about a year ago for around $1000, which isn't too shaby for a a switch that can do QoS, VLAN and PoE ... has a nice web interface too.
Current Netgear model is FSM7326P (not sure if its a new SKU since over a year ago when I got one).
http://www.netgear.com/products/details/FSM7326P.php$1000 might seem like a lot, but at ~$42 per port, its in the same ballpark of what you'd pay just for power injectors (not to mention its a layer 3 managed switch).
-AdamOn 21/04/06, Andrew Latham <[EMAIL PROTECTED]> wrote:
D-link has a nice one, optional 5 year warranty on some of thecommercial stuffOn 4/21/06, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:> Hi listers,> I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's.  I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office.  However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price.  I would appreciate any input people have to offer.
>> Thanks,>> James>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email!
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Steve Kennedy]

On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote:

> D-link has a nice one, optional 5 year warranty on some of the
> commercial stuff

Though beware, some of the D-Link ones only have half the ports with
PoE.


Steve


-- 
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UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[Andrew Latham]

D-link has a nice one, optional 5 year warranty on some of the
commercial stuff




On 4/21/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi listers,
> I am looking for people who have used Power over Ethernet switches, 
> primarily in conjunction with Polycom IP 501's.  I've been looking at the 
> Linksys SRW224P, since I've had good luck with the SRW224 in our office.  
> However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary 
> in price.  I would appreciate any input people have to offer.
>
> Thanks,
>
> James
>
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--
---
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[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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[Asterisk-Users] Power over Ethernet (PoE) switch recommendations

[james.texter]

Hi listers,
I am looking for people who have used Power over Ethernet switches, 
primarily in conjunction with Polycom IP 501's.  I've been looking at the 
Linksys SRW224P, since I've had good luck with the SRW224 in our office.  
However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in 
price.  I would appreciate any input people have to offer.

Thanks,

James

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Re: [Asterisk-Users] problem with TE205

[Ondrej Valousek]

Try "pri show span 1" and send me the result.

Augustine Olaifa wrote:
>
> Hello,
>
> I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have
> my jumper set (i.e closed to use the E1 facility.)
>
> but when i connect the E1 from my telco the LED on the TDM card is
> green and also when i look in zttool the status are ok.
>
> when i try to place a call out I get the following:
>
> -- Accepting AUTHENTICATED call from 196.1.178.172:
>> requested format = gsm,
>> requested prefs = (),
>> actual format = gsm,
>> host prefs = (g729|gsm),
>> priority = mine
> -- Executing Dial("IAX2/233066391-4", "Zap/g5/8105156,30") in new
> stack
> Apr 21 13:50:43 NOTICE[1565]: app_dial.c:1011 dial_exec_full: Unable
> to create channel of type 'Zap' (cause 0 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
> = Everyone is busy/congested at this time (1:0/0/1)
> -- Hungup 'IAX2/233066391-6'
>
> -- Accepting AUTHENTICATED call from 196.1.178.172:
>> requested format = gsm,
>> requested prefs = (),
>> actual format = gsm,
>> host prefs = (g729|gsm),
>> priority = mine
> -- Executing Dial("IAX2/233066391-4", "Zap/g5/8105156,30") in new
> stack
> Apr 21 14:28:52 NOTICE[1095]: app_dial.c:1011 dial_exec_full: Unable
> to create channel of type 'Zap' (cause 0 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
> pbx23*CLI>
>
>
> My config files:
> zaptel.conf
> ===
> span=1,1,0,ccs,hdb3
> span=2,1,0,ccs,hdb3
> bchan=1-15,17-30
> dchan=16
>
> bchan=31-45,47-60
> dchan=46
> loadzone = us
>
>
>
> zapata.conf
> 
>
> group=5
> context=outsideserver
> ;context= incoming
> switchtype=euroisdn
> pridialplan=national
> priindication=inband
> signalling=pri_cpe
> resetinterval=60
> overlapdial=yes
> callprogress=yes
> callerid=22004488
> ;channel=1-15,17-30
>
>
> bchan=1-15,17-31
> dchan=16
> txgain=3.0
> rxgain=3.0
>
>
>
>
> group=7
> context=weareout
> ;context= incoming
> switchtype=euroisdn
> ;context= incoming
> switchtype=euroisdn
> pridialplan=national
> priindication=inband
> signalling=pri_cpe
> resetinterval=60
> overlapdial=yes
> callprogress=yes
> callerid=155152
> ;channel=31-45
> ;channel=47-61
>
> bchan=31-45,47-60
> dchan=46
>
>
> txgain=3.0
> rxgain=3.0
>
> idledial=1551520
> [EMAIL PROTECTED]
> minunused=2
> minidle=1
>
> and this is my dialplan.trying to call out on group 5
>
> extensions.conf
> ==
> [outsideserver]
> include=> default
> include=> national
> include=> local
>
> [weareout]
> exten => 1551520,1,Answer
> ..
> ..
>
>
> [national]
> exten =>_02X.,1,Dial,Zap/g5/${EXTEN:2},30
> exten => _0802X.,1,Dial,Zap/g5/${EXTEN},30
> exten => _080[36]X.,1,Dial,Zap/g5/${EXTEN},30
> ;exten => _0804X.,1,Dial,Zap/g5/${EXTEN},30
>
> could this be a hardware problem? or is the TDM card defected?
>
>
> Please anyone ever had this problem i will like to share how it was
> sorted out.
>
>
> regards
> ==
>
> Olaifa Augustine
> General Data Engineering Services Ltd
> 18b oshin road,kongi bodija
> p.o.box 29460, secretariate,
> ibadan.
> tel:- 234-2-8105156
>
>
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[Asterisk-Users] problem with TE205

[Augustine Olaifa]

Hello,

I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my 
jumper set (i.e closed to use the E1 facility.)


but when i connect the E1 from my telco the LED on the TDM card is green 
and also when i look in zttool the status are ok.


when i try to place a call out I get the following:

-- Accepting AUTHENTICATED call from 196.1.178.172:
   > requested format = gsm,
   > requested prefs = (),
   > actual format = gsm,
   > host prefs = (g729|gsm),
   > priority = mine
-- Executing Dial("IAX2/233066391-4", "Zap/g5/8105156,30") in new 
stack
Apr 21 13:50:43 NOTICE[1565]: app_dial.c:1011 dial_exec_full: Unable to 
create channel of type 'Zap' (cause 0 - Unknown)

  == Everyone is busy/congested at this time (1:0/0/1)
= Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'IAX2/233066391-6'

-- Accepting AUTHENTICATED call from 196.1.178.172:
   > requested format = gsm,
   > requested prefs = (),
   > actual format = gsm,
   > host prefs = (g729|gsm),
   > priority = mine
-- Executing Dial("IAX2/233066391-4", "Zap/g5/8105156,30") in new 
stack
Apr 21 14:28:52 NOTICE[1095]: app_dial.c:1011 dial_exec_full: Unable to 
create channel of type 'Zap' (cause 0 - Unknown)

  == Everyone is busy/congested at this time (1:0/0/1)
pbx23*CLI>


My config files:
zaptel.conf
===
span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3
bchan=1-15,17-30
dchan=16

bchan=31-45,47-60
dchan=46
loadzone = us



zapata.conf


group=5
context=outsideserver
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=22004488
;channel=1-15,17-30


bchan=1-15,17-31
dchan=16
txgain=3.0
rxgain=3.0




group=7
context=weareout
;context= incoming
switchtype=euroisdn
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=155152
;channel=31-45
;channel=47-61

bchan=31-45,47-60
dchan=46


txgain=3.0
rxgain=3.0

idledial=1551520
[EMAIL PROTECTED]
minunused=2
minidle=1

and this is my dialplan.trying to call out on group 5

extensions.conf
==
[outsideserver]
include=> default
include=> national
include=> local

[weareout]
exten => 1551520,1,Answer
..
..


[national]
exten =>_02X.,1,Dial,Zap/g5/${EXTEN:2},30
exten => _0802X.,1,Dial,Zap/g5/${EXTEN},30
exten => _080[36]X.,1,Dial,Zap/g5/${EXTEN},30
;exten => _0804X.,1,Dial,Zap/g5/${EXTEN},30

could this be a hardware problem? or is the TDM card defected?


Please anyone ever had this problem i will like to share how it was sorted 
out.



regards
==

Olaifa Augustine
General Data Engineering Services Ltd
18b oshin road,kongi bodija
p.o.box 29460, secretariate,
ibadan.
tel:- 234-2-8105156


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Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT.

[Moises Silva]

A couple of weeks ago, libmfcr2 has a small error in the tone
signaling for the call setup, that was fixed 2 weeks ago or so,
please, wich version of libmfcr2 are you using? if you dont know try
upgrading to the latest version. Im pretty much sure that you have the
very same problem we had.

Regards

On 4/21/06, Jefferson Carvalho <[EMAIL PROTECTED]> wrote:
> Hello All,
>
> I'm facing problems with Unicall on this scenario :
>
> CentOS 4.3 - Running on x86_64
> Asterisk 1.2.7.1
> Zaptel 1.2.5
>
> When running zttool , shows all Spans OK.
>
> But I can't receive and make calls.
>
> I tried to change many parameters and still doesn't work.
>
> Any clues ?
>
> * unicall.conf
>
> [channels]
>
> language=br
>
> context=incoming-pstn
> usecallerid=yes
> hidecallerid=no
> immediate=no
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancellforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
>
> rxgain=0.0
> txgain=0.0
> faxdetect=both
> loglevel=255
> protocolclass=mfcr2
> protocolvariant=br,20,4
> protocolend=cpe
> group=1
> callgroup=1
>
> channel => 1-15
> channel => 17-31
> channel => 32-46
> channel => 48-62
> channel => 63-77
> channel => 94-108
> channel => 110-124
>
> * zaptel.conf *
>
> loadzone=br
> defaultzone=br
>
>
> span=1,1,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
>
> span=2,0,0,cas,hdb3
> cas=32-46:1101
> cas=48-62:1101
>
>
> span=3,0,0,cas,hdb3
> cas=63-77:1101
> cas=79-93:1101
>
> span=4,0,0,cas,hdb3
> cas=94-108:1101
> cas=110-124:1101
>
>
>
> * lor error *
>
> -- Executing Dial("SIP/1000-1de2", "Unicall/g1/40020022|40|Ttr") in new
> stack
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Call control(1)
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Make call
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Making a new call with CRN 32769
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 0001  ->  [1/   1/Idle  /Idle ]
> -- Called g1/40020022
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Dialing
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1  <-   [1/  40/Seize /Idle ]
> Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 4 on  ->  [2/  40/Group I   /Idle ]
> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 R2 prot. err. [2/  40/Group I   /DNIS ] cause
> 32769 - T1 timed out
> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 4 off ->  [1/   1/Idle  /Idle ]
> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Protocol failure
> -- Unicall/1 protocol error. Cause 32769
> Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Channel echo cancel
> Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Channel gains
> Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Channel switching
> -- Hungup 'UniCall/1-1'
>   == Everyone is busy/congested at this time (1:0/0/1)
>   == Auto fallthrough, channel 'SIP/1000-1de2' status is 'CHANUNAVAIL'
> Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1  <- 1011  [1/   1/Idle  /Idle ]
> Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
>
> Jefferson Carvalho
>
>
>
>
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Re: [Asterisk-Users] Polycom MWI

[Sean Cook]

Try specifing [EMAIL PROTECTED]  I know their have been some changes
with the implicit defining of the voicemail groupsthat may have
something to do with it... I didn't have to do anything special for my
polycoms.

Sean

On Fri, 2006-04-21 at 06:17 -0400, Andrew Kohlsmith wrote:
> On Friday 21 April 2006 00:28, Kerry Garrison wrote:
> > Didn't help. Could I be missing something else?
> 
> In Avi's footsteps, here is my phone.cfg and sip.conf entry.  This works for 
> 12 phones.  Note that I'm not subscribing to anything on the Polycom; 
> Asterisk sends MWI for the mailbox to the phone when there are messages 
> waiting, and the user can dial '999' to access voicemail.
> 
> 
>  msg.mwi.1.subscribe=""
> msg.mwi.1.callBackMode="contact"
> msg.mwi.1.callBack="999"
> />
> 
> 
> [211]
> context=polycom_outgoing
> type=friend
> host=dynamic
> secret=*
> disallow=all
> allow=ulaw
> mailbox=211
> vmexten=voicemail
> dtmfmode=rfc2833
> callgroup=1
> pickupgroup=1
> callerid=Jack <211>
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RE: [Asterisk-Users] Polycom MWI

[Kerry Garrison]

Thanks a ton!!

When using Extensions mode (the default) this would be:

[EMAIL PROTECTED]

When Using Users and Devices mode this would be:

[EMAIL PROTECTED]

Thanks for the guidance there, this has been driving me nuts.

-Kerry

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Bill Gibbs
> Sent: Friday, April 21, 2006 5:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Polycom MWI
> 
> Ohh yeah good point.  I had a similar issue when I started 
> using FreePBX and it didn't fill out the mailbox field 
> automatically.  Once I added the [EMAIL PROTECTED] there the MWI 
> started working as well.
> 
> Bill
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Avi Miller
> Sent: Friday, April 21, 2006 12:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom MWI
> 
> Kerry Garrison wrote:
> > Didn't help. Could I be missing something else?
> 
> My phone.cfg looks like this:
> 
>msg.mwi.1.subscribe="300"
> msg.mwi.1.callBackMode="contact"
> msg.mwi.1.callBack="*97"/>
> 
> And sip.conf for extension 300:
> 
> [300]
> username=300
> type=friend
> secret=***
> record_out=Adhoc
> record_in=Adhoc
> qualify=no
> port=5060
> pickupgroup=1
> nat=never
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=rfc2833
> disallow=all
> context=from-internal
> canreinvite=no
> callgroup=1
> callerid=Polycom IP501 <300>
> allow=alaw
> allow=g729
> 
> 
> Mine works fine, so I hope that helps. :)
> 
> --
> National Manager - Special Projects
> 
> < Sydney / Melbourne / Canberra / Hobart / London />
>2/340 Gore Street  T: +61 (0) 3 9486 0411
>Fitzroy, VIC   F: +61 (0) 3 9486 0611
>3065   W: http://www.squiz.net/
> 
> .>> Open Source  - Own it  -  Squiz.net ./> 
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[Asterisk-Users] Flash Panel / Queue Slots

[Thomas Broda]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello,

is there any way to make the Flash Operator Panel show which agents are
logged in in a specific queue? (both static and dynamic agents)

I've played around with the queue / queue agents settings from the Flash
Panel documentation (http://www.asternic.org). The way it is described
there, I could only make the Flash panel show that a queue 8in general)
received a call from a specific extension.

- --
Thomas Broda, Systemadministration Frankfurt
FIRSTGATE AG,Im MediaPark 5, 50670 Koeln
Telefon: +49 (0) 2 21 / 45 45-747
Telefax: +49 (0) 2 21 / 45 45-710
Internet: http://www.firstgate.de
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2.2 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFESOZZulxz1xno2o4RApgqAJ99LdLeExyRPPCOTMkl/qGOWYW7XwCdFRRQ
Nsg9vqxFtilaPaeqwSn1v3I=
=YV2A
-END PGP SIGNATURE-
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[Asterisk-Users] Re: Modem connection

[Tomislav Parčina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> By luck, maybe. The only solution that looks like it should be fairly 
> solid is V.150, and I've only seen that on Cisco boxes so far.

Hi Steve!

Can you tell me more about Cisco box that are you talking about?


--
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Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

Hi Klaus,

> Thanks for the detailed answers and isdn for Linux basics.  I will take
> the opportunity to ask some more questions :-)
> 
> On Fri, April 21, 2006 12:24, Armin Schindler said:
> > On Fri, 21 Apr 2006, Klaus Darilion wrote:
> >> 3. Is the PIN layout for TE or NT mode?
> >
> > It is TE PIN layout, you need a crossed cable with 100Ohm termination for
> > NT-mode.
> 
> Why do I need a cable with 100Ohm termination? Shouldn't be the
> termination inside the DIVA Server? Until now (with quadbri and other isdn
> card) I only used CAT5 cables with BRI-crossover PIN layout. No resistors.
> Can you please explain this a little bit more or give me links to the
> wiring basics?

I must admit that I don't really know that. Maybe the quadbri has this 
termination automatically on board. Since the Eicon DIVA Server card has 
basically a TE port, the termination is necessary.
I always use the 100Ohm termination in my NT-cross-cables.

Here is a short description on Melware Wiki:
  http://www.melware.org/BriCrossCable
 
> >> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
> >> is the
> >> better (more expensive) card which also offers FAX on/offramp.
> >> Nevertheless I
> >> can use V-4BRI for faxing when using spandsp. Both cards do support
> >> onboard
> >> echo cancellation. Are this assumptions correct?
> >
> > Yes.
> 
> This means all kind of B-channel data (human voice, modem, fax, data ...)
> can be handled in pass-through also with the V-4BRI card?

Yes. In bridge mode (CAPI calls that Line-Interconnect) the b-channels are 
connected together, no matter what kind of data is going through.
 
> >> - For the hardware part there are also 2 choices. Either use the Eicon
> >> drivers
> >> included in 2.6 and divactrl or use the source packages from eicon and
> >> the
> >> tools included. Here I'm not sure which method is better. Further I do
> >> not
> >> know how this is related with isdn4linux or other linux ISDN stuff.
> >> From:
> >> http://www.eicon.com/support/helpweb/slnxen/asterisk.asp
> >
> > The in-kernel driver in 2.6 is the so called v2 driver from Melware. This
> > driver works very good. But the driver from Eicons sourceRPM (Melware
> > calls
> > it v3) is the newer one with many more features, more supported cards,
> > newer
> > firmware... (like RTP support which is used with newer chan-capi-cm as
> > well).
> 
> On the Eicon homepage I find to source packages: version 7.7 (they call
> ist stable) and version 8.0 (they call it beta). To which one of these do
> you refer with "v3"? Or is there another version somewhere hidden on the
> homepage?

Melware's v3 references both. v3 is just the version of the drivers/firmware 
compared with the v2 which is the status of the driver in kernel 2.6.
But please don't be confused with this v3 naming. The Eicon source package
(7.7 or 8.0) is very good.
Melware will create a v3 driver package out of that driver status including
some open-source community changes/add-ons. But this is not finished yet.
 
> What is meant with "newer firmware"? Is there a new firmware available
> which must be flashed into the cards?

Not flashed, but loaded. When the Eicon DIVA Server cards are started, the 
tool divactrl will activate the card by loading the firmware automatically 
and configuring it (the ports) according to settings.
New firmware means: new features has been added!
Remember, these cards are active cards. A system is running on that card 
handling the ISDN protocol (the hosts CPU/the driver does not need to care 
about that) as well as the b-channel data processing for echo-cancel, fax, 
modem, RTP, etc.
 
> Is the new v3 ready to use in production environment?

As written above the Melware's v3 is not finished yet, but the Eicons source 
package is definately ready for that (we have it running in many production 
systems with different applications).

Also, if you find any problem or missing feature, we can help you very fast 
since we work very closely with Eicon. And even there is a change in the 
firmware needed, Eicons response time is excellent.

Armin
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Re: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??

[Shane Young]

Quoting Pimjai Wesnarat <[EMAIL PROTECTED]>:

> Hi,
>
> I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
> It works fine when I have just 1 voice installed. Now I have 2 voices in
> the same language installed but I can't seem to find the way to select
> which voice to use in Swift's application in Asterisk. Does anyone know??

[cepstral-demo]
exten => s,1,Answer
exten => s,n,wait(1)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Duchess.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Walter.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Shouty.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is William.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Whispery.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Robin.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Linda.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Emily.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Diane.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is David.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Duncan.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Damien.)
exten => s,n,Cepstral(Hello and welcome to the world of 
text to speech using
Cepstral.  My name is Callie.)
exten => s,n,Cepstral(Hello and welcome to the world of text 
to speech using
Cepstral.  My name is Dog.)
exten => s,n,Cepstral(Hello and welcome to the world of text 
to speech using
Cepstral.  My name is Amy.)


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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Avi Miller wrote:
> Armin Schindler wrote:
> > > Using spandsp and V-4BRI does not work?
> > 
> > That will work. It's just that the on-board fax capabilities won't work,
> > but any other software fax will work like with other cards.
> 
> Just a note that I've never managed to get this to work on my V-4BRI cards: If
> I attempt to use SpanDSP to send or receive a fax, Asterisk will crash. This
> happens on multiple servers, so now I don't even bother compiling SpanDSP
> support onto my BRI-only Asterisk servers.
> 
> If anyone knows how to actually get this working, I'm all ears.

I don't know much about spandsp, but a crash should not happen.
Since Asterisk crashes, the problem is not the divas driver. It could be 
chan-capi, but I don't why this would make a difference, because for 
chan-capi it is voice data just like with any other voice connection.

Can you provide a backtrace to track this problem down?

Armin

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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Klaus Darilion wrote:
> Hi!
> 
> I've forgotten to ask an important question:
> 
> Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?

Yes, and each port can be configured separately.
 
> thanks
> klaus
> 
> btw: we should collect Q&A somewhere on a Wiki.

Yes. For that purpose Melware has activated melware.org

Armin
 
> On Fri, April 21, 2006 12:33, David Waugh said:
> > Hello Klaus,
> >
> > Normally, Diva Server adapters are operated as terminal equipment. In
> > this case, they derive their timing from the signal received from the
> > NT, for example PSTN or PBX, and use this derived timing to synchronize
> > their transmitted signal. If you use the Diva Server adapter as network
> > termination, it generates the timing from which the terminal equipment
> > derives its timing and synchronization.
> >
> > Thus TE=Slave
> > And NT=Master
> >
> > I haven't tried spandsp and a V-Series card as the card was not designed
> > for fax applications. As a result it has never been tested. If you want
> > to fax with the card then we would recommend a normal Diva Server card.
> >
> > It would probably work in a pass through scenario as just audio is being
> > relayed by the card. However, I have not tested this and could not
> > comment if this will work with a V-BRI card.
> >
> > Bascially if you are looking to do pure voice, then use a V-Series card.
> > If you want to do anything more than voice, then use a normal Diva
> > Server card.
> >
> > Kind regards
> > David
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Klaus
> > Darilion
> > Sent: 21 April 2006 11:04
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions
> >
> > On Fri, April 21, 2006 11:45, David Waugh said:
> >> Hello Klaus,
> >>
> >> I will answer your questions In turn:
> >>
> >> 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
> >> Yes they do.
> >>
> >> 2. Can the clock (master/slave) be configured independent from the
> > mode
> >> (TE/NT)
> >> No
> >
> > Thus, what is the limitation?
> > TE = slave  ?
> > NT = master ?
> >
> >
> >> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
> >> is the better (more expensive) card which also offers FAX on/offramp.
> >> Nevertheless I can use V-4BRI for faxing when using spandsp. Both
> > cards
> >> do support onboard echo cancellation. Are this assumptions correct?
> >>
> >> The difference between the Diva Server V-Series and the normal Diva
> >> Server cards is that you can only use the V-Series for Voice
> >> applications. You can not fax with a V-Series card. If you try and fax
> >> with a V-Series card you will get an error.
> >
> > Using spandsp and V-4BRI does not work?
> >
> > What about a pass-through scenario:
> >
> > PSTN <-2xBRI-> Asterisk <-2xBRI-> oldPBX
> >
> > Does Asterisk with the V-4BRI allows to pass through fax calls (or other
> > data calls) from the PSTN to the old PBX?
> >
> >
> >> 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
> >> I'm a little bit confused about what I have to install.
> >>
> >> This is just a package that contains the capi.h file that is needed to
> >> compile the chan_capi-cm driver.
> >
> > btw: libcapi20-dev on debian ;-)
> >
> > regards
> > klaus
> >
> >
> >
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> 
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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Avi Miller]

Armin Schindler wrote:

Using spandsp and V-4BRI does not work?


That will work. It's just that the on-board fax capabilities won't work, but 
any other software fax will work like with other cards.


Just a note that I've never managed to get this to work on my V-4BRI 
cards: If I attempt to use SpanDSP to send or receive a fax, Asterisk 
will crash. This happens on multiple servers, so now I don't even bother 
compiling SpanDSP support onto my BRI-only Asterisk servers.


If anyone knows how to actually get this working, I'm all ears.

cYa,
Avi

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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Klaus Darilion wrote:
> On Fri, April 21, 2006 12:33, David Waugh said:
> > Hello Klaus,
> ...
> > I haven't tried spandsp and a V-Series card as the card was not designed
> > for fax applications. As a result it has never been tested. If you want
> > to fax with the card then we would recommend a normal Diva Server card.
> >
> > It would probably work in a pass through scenario as just audio is being
> > relayed by the card. However, I have not tested this and could not
> > comment if this will work with a V-BRI card.
> >
> > Bascially if you are looking to do pure voice, then use a V-Series card.
> > If you want to do anything more than voice, then use a normal Diva
> > Server card.
> 
> Hi David!
> 
> You never know which kind of voice (human, modem, fax) will pass through
> the B channel. I guess handling pass-trough modem calls should work, if
> they card will disable the echo canceller on detection of the "special
> tones" used in modem conversations. Do they?

The echo-chanceler can do that, but chan-capi disables echo-cancel on bridge 
anyway. So on bridge no data is modified.

Also the automatic disable of echocancel on e.g. a fax-tone when using a 
software fax can be configured as option in capi.conf of chan-capi.

Armin

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RE: [Asterisk-Users] Polycom MWI

[Bill Gibbs]

Ohh yeah good point.  I had a similar issue when I started using FreePBX
and it didn't fill out the mailbox field automatically.  Once I added
the [EMAIL PROTECTED] there the MWI started working as well.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller
Sent: Friday, April 21, 2006 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom MWI

Kerry Garrison wrote:
> Didn't help. Could I be missing something else?

My phone.cfg looks like this:

  

And sip.conf for extension 300:

[300]
username=300
type=friend
secret=***
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
pickupgroup=1
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-internal
canreinvite=no
callgroup=1
callerid=Polycom IP501 <300>
allow=alaw
allow=g729


Mine works fine, so I hope that helps. :)

-- 
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< Sydney / Melbourne / Canberra / Hobart / London />
   2/340 Gore Street  T: +61 (0) 3 9486 0411
   Fitzroy, VIC   F: +61 (0) 3 9486 0611
   3065   W: http://www.squiz.net/

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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[Klaus Darilion]

Hi!

I've forgotten to ask an important question:

Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode?

thanks
klaus

btw: we should collect Q&A somewhere on a Wiki.

On Fri, April 21, 2006 12:33, David Waugh said:
> Hello Klaus,
>
> Normally, Diva Server adapters are operated as terminal equipment. In
> this case, they derive their timing from the signal received from the
> NT, for example PSTN or PBX, and use this derived timing to synchronize
> their transmitted signal. If you use the Diva Server adapter as network
> termination, it generates the timing from which the terminal equipment
> derives its timing and synchronization.
>
> Thus TE=Slave
> And NT=Master
>
> I haven't tried spandsp and a V-Series card as the card was not designed
> for fax applications. As a result it has never been tested. If you want
> to fax with the card then we would recommend a normal Diva Server card.
>
> It would probably work in a pass through scenario as just audio is being
> relayed by the card. However, I have not tested this and could not
> comment if this will work with a V-BRI card.
>
> Bascially if you are looking to do pure voice, then use a V-Series card.
> If you want to do anything more than voice, then use a normal Diva
> Server card.
>
> Kind regards
> David
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Klaus
> Darilion
> Sent: 21 April 2006 11:04
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions
>
> On Fri, April 21, 2006 11:45, David Waugh said:
>> Hello Klaus,
>>
>> I will answer your questions In turn:
>>
>> 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
>> Yes they do.
>>
>> 2. Can the clock (master/slave) be configured independent from the
> mode
>> (TE/NT)
>> No
>
> Thus, what is the limitation?
> TE = slave  ?
> NT = master ?
>
>
>> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
>> is the better (more expensive) card which also offers FAX on/offramp.
>> Nevertheless I can use V-4BRI for faxing when using spandsp. Both
> cards
>> do support onboard echo cancellation. Are this assumptions correct?
>>
>> The difference between the Diva Server V-Series and the normal Diva
>> Server cards is that you can only use the V-Series for Voice
>> applications. You can not fax with a V-Series card. If you try and fax
>> with a V-Series card you will get an error.
>
> Using spandsp and V-4BRI does not work?
>
> What about a pass-through scenario:
>
> PSTN <-2xBRI-> Asterisk <-2xBRI-> oldPBX
>
> Does Asterisk with the V-4BRI allows to pass through fax calls (or other
> data calls) from the PSTN to the old PBX?
>
>
>> 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
>> I'm a little bit confused about what I have to install.
>>
>> This is just a package that contains the capi.h file that is needed to
>> compile the chan_capi-cm driver.
>
> btw: libcapi20-dev on debian ;-)
>
> regards
> klaus
>
>
>
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Re: [Asterisk-Users] Real-time Database Front-end

[Rich Adamson]

James Nunnerley wrote:
I’ve had Asterisk working on a test platform really well, but I’ve never 
found a decent web front end, that works in real-time.


 

I’ve got a couple of incoming numbers that I’d like to have some IVR on 
(i.e. select this option etc), and then distribute the calls 
appropriately to various SIP agents, but also in some cases back out to 
a PSTN/Mobile number.


 

I have this working well on a flat file config, but would like to get it 
in a db...


 

I’ve taken a look at [EMAIL PROTECTED] however I want to install it on a current 
server, running Fedora Core 4, and the only option I can see is 
installing it as a new OS – or not on FC4.


 


Can anyone provide any advice?


http://asteriskathome.sourceforge.net/handbook/index.html

Section 2.3 says...

mkdir /var/aah_load
cp asteriskathome-1.5.tar.gz /var/aah_load
cd /var/aah_load
tar xvfz asteriskathome-1.5.tar.gz
./install.sh

Change the 1.5 to whatever is current; should work.

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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Klaus Darilion]

Hi Armin!

Thanks for the detailed answers and isdn for Linux basics.  I will take
the opportunity to ask some more questions :-)

On Fri, April 21, 2006 12:24, Armin Schindler said:
> On Fri, 21 Apr 2006, Klaus Darilion wrote:
>> 3. Is the PIN layout for TE or NT mode?
>
> It is TE PIN layout, you need a crossed cable with 100Ohm termination for
> NT-mode.

Why do I need a cable with 100Ohm termination? Shouldn't be the
termination inside the DIVA Server? Until now (with quadbri and other isdn
card) I only used CAT5 cables with BRI-crossover PIN layout. No resistors.
Can you please explain this a little bit more or give me links to the
wiring basics?

>> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
>> is the
>> better (more expensive) card which also offers FAX on/offramp.
>> Nevertheless I
>> can use V-4BRI for faxing when using spandsp. Both cards do support
>> onboard
>> echo cancellation. Are this assumptions correct?
>
> Yes.

This means all kind of B-channel data (human voice, modem, fax, data ...)
can be handled in pass-through also with the V-4BRI card?

>> - For the hardware part there are also 2 choices. Either use the Eicon
>> drivers
>> included in 2.6 and divactrl or use the source packages from eicon and
>> the
>> tools included. Here I'm not sure which method is better. Further I do
>> not
>> know how this is related with isdn4linux or other linux ISDN stuff.
>> From:
>> http://www.eicon.com/support/helpweb/slnxen/asterisk.asp
>
> The in-kernel driver in 2.6 is the so called v2 driver from Melware. This
> driver works very good. But the driver from Eicons sourceRPM (Melware
> calls
> it v3) is the newer one with many more features, more supported cards,
> newer
> firmware... (like RTP support which is used with newer chan-capi-cm as
> well).

On the Eicon homepage I find to source packages: version 7.7 (they call
ist stable) and version 8.0 (they call it beta). To which one of these do
you refer with "v3"? Or is there another version somewhere hidden on the
homepage?

What is meant with "newer firmware"? Is there a new firmware available
which must be flashed into the cards?

Is the new v3 ready to use in production environment?

thanks
klaus



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[Asterisk-Users] Airspan / Arelnet GW and Asterisk

[Michel Luczak]

Hi allHas anyone seen this kind of messages : Apr 21 12:04:12 NOTICE[89928]: chan_sip.c:3449 process_sdp: Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp'I get this using a priorietary (Airspan's prime, ex-arelnet) E1 gateway with asterisk. It seems like the SIP protocol they're using is somehow weird...Is there a configuration-level solution or should i patch asterisk so that it ignores the trailing multipart content of the INVITE request ?(for the record, the end of the request looks like :a=rtpmap:0 PCMU/8000a=ptime:40a=rtpmap:17 T38/8000a=ptime:40a=ptime:20--unique-boundary-1Content-Type: application/x-Arelnet;Object=OrigTrunkLNo=1--unique-boundary-1- (11 headers 26 lines)---) -- Michel Luczak[EMAIL PROTECTED]***Internet Email Confidentiality Footer**Privileged/Confidential Information may be contained in this message.If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone.In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind.  Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it.*** ___
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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[Klaus Darilion]

On Fri, April 21, 2006 12:33, David Waugh said:
> Hello Klaus,
...
> I haven't tried spandsp and a V-Series card as the card was not designed
> for fax applications. As a result it has never been tested. If you want
> to fax with the card then we would recommend a normal Diva Server card.
>
> It would probably work in a pass through scenario as just audio is being
> relayed by the card. However, I have not tested this and could not
> comment if this will work with a V-BRI card.
>
> Bascially if you are looking to do pure voice, then use a V-Series card.
> If you want to do anything more than voice, then use a normal Diva
> Server card.

Hi David!

You never know which kind of voice (human, modem, fax) will pass through
the B channel. I guess handling pass-trough modem calls should work, if
they card will disable the echo canceller on detection of the "special
tones" used in modem conversations. Do they?

regards
klaus

>
> Kind regards
> David
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Klaus
> Darilion
> Sent: 21 April 2006 11:04
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions
>
> On Fri, April 21, 2006 11:45, David Waugh said:
>> Hello Klaus,
>>
>> I will answer your questions In turn:
>>
>> 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
>> Yes they do.
>>
>> 2. Can the clock (master/slave) be configured independent from the
> mode
>> (TE/NT)
>> No
>
> Thus, what is the limitation?
> TE = slave  ?
> NT = master ?
>
>
>> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
>> is the better (more expensive) card which also offers FAX on/offramp.
>> Nevertheless I can use V-4BRI for faxing when using spandsp. Both
> cards
>> do support onboard echo cancellation. Are this assumptions correct?
>>
>> The difference between the Diva Server V-Series and the normal Diva
>> Server cards is that you can only use the V-Series for Voice
>> applications. You can not fax with a V-Series card. If you try and fax
>> with a V-Series card you will get an error.
>
> Using spandsp and V-4BRI does not work?
>
> What about a pass-through scenario:
>
> PSTN <-2xBRI-> Asterisk <-2xBRI-> oldPBX
>
> Does Asterisk with the V-4BRI allows to pass through fax calls (or other
> data calls) from the PSTN to the old PBX?
>
>
>> 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
>> I'm a little bit confused about what I have to install.
>>
>> This is just a package that contains the capi.h file that is needed to
>> compile the chan_capi-cm driver.
>
> btw: libcapi20-dev on debian ;-)
>
> regards
> klaus
>
>
>
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Re: [Asterisk-Users] Redirecting to another service/server

[broadbandvoice]

Did you get an answer to this? I am interested in SIP to SIP calls on other networks thereby by-passing the pstn.
 
-- Original message -- From: Nick Hoffman <[EMAIL PROTECTED]> > Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? > > For instance, an extension behind Asterisk dials 99751234, and Asterisk > says "that starts with 99. let's strip off the 99 and call 751234 at FWD, > IE: sip:[EMAIL PROTECTED]:5060". > > Is that possible, or would services such as FWD reject the call since the > device making the call (Asterisk) hasn't registered? > > Thanks! > -- Nick > e: [EMAIL PROTECTED] > p: +61 7 5591 3588 > f: +61 7 5591 6588 > > If you receive this email by mistake, please notify us and do not make any > use of the email. We do not waive any privilege, confidentiality or > copyright associated wi
 th it.
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RE: [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6

[Steve Totaro]

Cubix has always crashed on me while using moderately.  Nice looking phone but 
not stable.

Idefisk works great.



On 20 Apr 2006, at 16:09, Peter Beckman wrote:

> I've tried Idefisk and Cubix Softphones, and they both work fine, 
> except
> for two issues:
>
> 1. Idefisk seems to have a longer delay between the time I can hit
>tones, and
>
> 2. Cubix, while can send DTMF faster, never actually connects 
> to an
>Asterisk-dialed call -- I can't hear the party who answers.
>
> #2 has been asked but unanswered here:
>
> http://lists.digium.com/pipermail/asterisk-users/2006-February/
> 139240.html
>
> I've got a weird problem with both Firefly & iaxLite (both IAX
> softphones).  They don't seem to stop ringing when an incoming 
> call is
> make to them.  If the call is answered the conversation starts 
> both ways
> but the ringing sound still keeps going and the softphones keep
> displaying that a call is coming in (but they do not display 
> that the
> call is answered).
>
> I read on the voip-info website that the "fix" for this with 
> Firefly is
> to set jitterbuffer to no  which I tried but it didn't work.
>
> Because the problem is with two IAX softphones I'm not sure 
> whether its
> a configuration problem with the asterisk server or, by change, 
> the same
> bug with both softphones.
>
> Has anyone else come up against this?
>
> Can you change the amount of time between DTMF in Idefisk?
>
> Can you modify a config to get Cubix to actually connect to a Dial()ed
> call?

Peter, do you have any packet logs (either ethereal or iax2 debug)
of the non-working IAX Dial() I can look at?
I may be able to diagnose the problem from that...

T.

Tim Panton
[EMAIL PROTECTED]



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RE: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swiftapplication??

[Steve Totaro]

Type "swift" at the command line so you can see the -options.  Then modify the 
line to use the correct switch and specify the name of the voice you want to 
use.
 
Thanks,
Steve

-Original Message- 
From: Pimjai Wesnarat [mailto:[EMAIL PROTECTED] 
Sent: Fri 4/21/2006 6:59 AM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's 
Swiftapplication??



Hi,

I'm using Cepstral as a TTS Engine for Asterisk with Swift application.
It works fine when I have just 1 voice installed. Now I have 2 voices in
the same language installed but I can't seem to find the way to select
which voice to use in Swift's application in Asterisk. Does anyone 
know??

Thank you,

Pim
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RE: [Asterisk-Users] Asterisk on Red Hat AS 4?

[Steve Totaro]

Red Hat AS 4 is the same as CentOS4X (with the Red Hat references stripped) and 
Asterisk works just fine on it.

-Original Message- 
From: Mimmus [mailto:[EMAIL PROTECTED] 
Sent: Fri 4/21/2006 6:59 AM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: [Asterisk-Users] Asterisk on Red Hat AS 4?



Hi,
I'm planning to install a new Asterisk server with a Digium TE410P card.
Can I use Red Hat Advanced Server 4 (latest update)?
Is this a good choice?
Is recompiling Asterisk simple with kernel 2.6?

Thanks
--
Domenico Viggiani

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[Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??

[Pimjai Wesnarat]

Hi,

I'm using Cepstral as a TTS Engine for Asterisk with Swift application. 
It works fine when I have just 1 voice installed. Now I have 2 voices in 
the same language installed but I can't seem to find the way to select 
which voice to use in Swift's application in Asterisk. Does anyone know??


Thank you,

Pim
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[Asterisk-Users] Asterisk on Red Hat AS 4?

[Mimmus]

Hi,
I'm planning to install a new Asterisk server with a Digium TE410P card.
Can I use Red Hat Advanced Server 4 (latest update)?
Is this a good choice?
Is recompiling Asterisk simple with kernel 2.6?

Thanks
-- 
Domenico Viggiani

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RE: [Asterisk-Users] AAH or Fedora an Asterisk by sources

[Steve Totaro]

Install from sources.  Run only Asterisk on the production machine, no extra 
processes.  If you must have databases, GUI, AGI and ... put those on another 
machine and call them across the network.  Also, I am not sure how much it 
helps, but eliminate asterisk modules you will not use by using noload= in 
modules.conf.  chmod 444 the files that you do not want to overwrite on the 
production box (zapata.conf, zaptel.conf, xxx_addtional, ...)
 
If you like [EMAIL PROTECTED], install it on a test machine.  Make all of your 
configs there and then copy the .confs to the production server (I use WinSCP 
to copy and also keep a set of copies in Source Safe.)
 
Thanks,
Steve  

-Original Message- 
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Fri 4/21/2006 6:28 AM 
To: asterisk-users@lists.digium.com 
Cc: 
Subject: [Asterisk-Users] AAH or Fedora an Asterisk by sources



Hello list users, I come to you in order to ask you your best 
recommendation for a large scale production Server, which Hill be your best 
recommendation between [EMAIL PROTECTED] and an installation from scratch with 
Fedora Core 4 and asterisk compiled by sources??

 

 

Thanks in advance, have fun!

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[Asterisk-Users] record_in / record_out configuration parameters

[Florian Muellner]

Hi all,
having performance problems with various SIP-Phones, the manufacturer 
adviced us to add these parameters in sip.conf - unfortunately, neither 
one of us has an idea what these are supposed to do.
I've seen various configuration files (sip.conf, iax.conf) posted on the 
net or this list using said paramters, but they seem to completely lack 
documentation (or is it just me?).
Grepping for them in the asterisk sources (and those of related 
packages) didn't show up anything either. We're running asterisk 1.2, 
just in case they're old parameters which have been removed or more 
recently added ones...
Could someone in the know give a short explanation of what these 
actually do?


Thanx in advance,

Flo
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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[David Waugh]

Hello Klaus,

Normally, Diva Server adapters are operated as terminal equipment. In
this case, they derive their timing from the signal received from the
NT, for example PSTN or PBX, and use this derived timing to synchronize
their transmitted signal. If you use the Diva Server adapter as network
termination, it generates the timing from which the terminal equipment
derives its timing and synchronization.

Thus TE=Slave
And NT=Master

I haven't tried spandsp and a V-Series card as the card was not designed
for fax applications. As a result it has never been tested. If you want
to fax with the card then we would recommend a normal Diva Server card.

It would probably work in a pass through scenario as just audio is being
relayed by the card. However, I have not tested this and could not
comment if this will work with a V-BRI card.

Bascially if you are looking to do pure voice, then use a V-Series card.
If you want to do anything more than voice, then use a normal Diva
Server card.

Kind regards
David


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus
Darilion
Sent: 21 April 2006 11:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions

On Fri, April 21, 2006 11:45, David Waugh said:
> Hello Klaus,
>
> I will answer your questions In turn:
>
> 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?
> Yes they do.
>
> 2. Can the clock (master/slave) be configured independent from the
mode
> (TE/NT)
> No

Thus, what is the limitation?
TE = slave  ?
NT = master ?


> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
> is the better (more expensive) card which also offers FAX on/offramp.
> Nevertheless I can use V-4BRI for faxing when using spandsp. Both
cards
> do support onboard echo cancellation. Are this assumptions correct?
>
> The difference between the Diva Server V-Series and the normal Diva
> Server cards is that you can only use the V-Series for Voice
> applications. You can not fax with a V-Series card. If you try and fax
> with a V-Series card you will get an error.

Using spandsp and V-4BRI does not work?

What about a pass-through scenario:

PSTN <-2xBRI-> Asterisk <-2xBRI-> oldPBX

Does Asterisk with the V-4BRI allows to pass through fax calls (or other
data calls) from the PSTN to the old PBX?


> 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
> I'm a little bit confused about what I have to install.
>
> This is just a package that contains the capi.h file that is needed to
> compile the chan_capi-cm driver.

btw: libcapi20-dev on debian ;-)

regards
klaus



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RE: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Klaus Darilion wrote:
> On Fri, April 21, 2006 11:45, David Waugh said:
> > 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI
> > is the better (more expensive) card which also offers FAX on/offramp.
> > Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards
> > do support onboard echo cancellation. Are this assumptions correct?
> >
> > The difference between the Diva Server V-Series and the normal Diva
> > Server cards is that you can only use the V-Series for Voice
> > applications. You can not fax with a V-Series card. If you try and fax
> > with a V-Series card you will get an error.
> 
> Using spandsp and V-4BRI does not work?

That will work. It's just that the on-board fax capabilities won't work, but 
any other software fax will work like with other cards.
 
> What about a pass-through scenario:
> 
> PSTN <-2xBRI-> Asterisk <-2xBRI-> oldPBX

No Problem! I have this running with 2 4BRI cards:
 PSTN <-4xBRI-> Asterisk/OpenPBX <-4xBRI NT-mode-> old PBX

> Does Asterisk with the V-4BRI allows to pass through fax calls (or other
> data calls) from the PSTN to the old PBX?

Yes, but bridge=yes should be used to pass trough without Asterisk/CPU. 
 
> > 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
> > I'm a little bit confused about what I have to install.
> >
> > This is just a package that contains the capi.h file that is needed to
> > compile the chan_capi-cm driver.
> 
> btw: libcapi20-dev on debian ;-)

exactly!

Armin

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[Asterisk-Users] AAH or Fedora an Asterisk by sources

[asterisk]

Hello list users, I come to you in order to ask you
your best recommendation for a large scale production Server, which Hill be
your best recommendation between [EMAIL PROTECTED] and an installation from scratch
with Fedora Core 4 and asterisk compiled by sources??

 

 

Thanks in advance, have fun!






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Re: [Asterisk-Users] some EICON Diva 4BRI questions

[Armin Schindler]

On Fri, 21 Apr 2006, Klaus Darilion wrote:
> Hi!
> 
> I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to
> have (short) answered before buying the DIVA card. I know there are several
> Eicon guys active on the list, thus I ask on the list instead of directly to
> Eicon so that all other will benefit as well.

I try to answer ;-)
 
> 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode?

Yes.
 
> 2. Can the clock (master/slave) be configured independent from the mode
> (TE/NT)

I don't know that one. But I can contact Eicon here.
 
> 3. Is the PIN layout for TE or NT mode?

It is TE PIN layout, you need a crossed cable with 100Ohm termination for 
NT-mode.
 
> 4. When I change the mode (implies 'YES' for question 1), will this also
> change the polarity on the connectors or do I have to use a BRI-crossover
> cable?

See answer on 3., cross cable is needed.
 
> 5. If a call is bridged from BRI->BRI, is it done directly on the BRI card or
> via Asterisk

You can configure this in the capi.conf of chan-capi-cm.
If you set bridge=yes, then the b-channels will be bridged on the card, not 
via Asterisk/CPU. This also works between different cards via Bus-Master 
DMA.
 
> 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the
> better (more expensive) card which also offers FAX on/offramp. Nevertheless I
> can use V-4BRI for faxing when using spandsp. Both cards do support onboard
> echo cancellation. Are this assumptions correct?

Yes.
 
> Usage in Asterisk: please correct me if I'm wrong:
> - Communication between the DIVA cards and Asterisk always happens via the
> CAPI interfaces.

Correct.
 
> - For the Asterisk part I have the choice of chan_misdn or chan_capi(-cm).
> After reading I come to the conclusion that chan_capi(-cm) should work better.

When using Eicon DIVA Server, you can use chan-capi only. chan-misdn is for 
passive isdn cards like the DIVA client cards.
 
> - For the hardware part there are also 2 choices. Either use the Eicon drivers
> included in 2.6 and divactrl or use the source packages from eicon and the
> tools included. Here I'm not sure which method is better. Further I do not
> know how this is related with isdn4linux or other linux ISDN stuff. From:
> http://www.eicon.com/support/helpweb/slnxen/asterisk.asp

The in-kernel driver in 2.6 is the so called v2 driver from Melware. This 
driver works very good. But the driver from Eicons sourceRPM (Melware calls 
it v3) is the newer one with many more features, more supported cards, newer 
firmware... (like RTP support which is used with newer chan-capi-cm as 
well).
 
> 1. "Ensure that you do not have the ISDN4Linux driver or the HiSax driver
> installed."   What is the ISDN4Linux driver? Is is something generic in Linux
> or do they talk about a certain ISDN4Linux driver for Eicon cards?

ISDN4Linux is the 'old' ISDN core of the Linux kernel. It is still there, 
but not developed any more. CAPI is used more and more. The HiSax driver as 
well as older Eicon driver in kernel uses this core. HiSax is the old driver 
for passive ISDN cards, mISDN is the new one here.
With CAPI and Eicon DIVA Server you don't need ISDN4Linux, but it should not
harm if installed in parallel, because CAPI can co-exist with it.
We (Melware) do provide a special driver where the new Eicon drivers can be 
used with ISDN4Linux as well. E.g. when you want to use the AT-emulator 
(ttyI) interfaces of ISDN4Linux with the Eicon DIVA Server cards. But Eicon 
does provide an own special tty-AT-emulator as well.
All these interfaces (CAPI, ISDN4Linux, tty-AT) can be used at the same time
if wanted.
 
> 2. " ...install the isdn4k-utils-devel package...". What is isdn4k ?
> I'm a little bit confused about what I have to install.

The isdn4k-utils (isdn for kernel utilities) contain the CAPI library 
(libcapi20) which is needed for chan-capi to compile/use. That is the reason 
for that package. Just make sure that you have the libcapi20.* and the 
header files of it (some distributions provide the header files via a 
special -devel package only).
Since I have created a special version of that libcapi20 to support 
remote-CAPI via TCP (ISDN Hardware in one maschine, Application in another), 
Melware will provide more packages via chan-capi.org / melware.org soon.

If you have further questions, please don't hesitate to ask.

Armin

--
Cytronics & Melware

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Re: [Asterisk-Users] Polycom MWI

[Andrew Kohlsmith]

On Friday 21 April 2006 00:28, Kerry Garrison wrote:
> Didn't help. Could I be missing something else?

In Avi's footsteps, here is my phone.cfg and sip.conf entry.  This works for 
12 phones.  Note that I'm not subscribing to anything on the Polycom; 
Asterisk sends MWI for the mailbox to the phone when there are messages 
waiting, and the user can dial '999' to access voicemail.





[211]
context=polycom_outgoing
type=friend
host=dynamic
secret=*
disallow=all
allow=ulaw
mailbox=211
vmexten=voicemail
dtmfmode=rfc2833
callgroup=1
pickupgroup=1
callerid=Jack <211>
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[Asterisk-Users] Unicall MFRC2 Problems with BrT.

[Jefferson Carvalho]

Hello All,

I'm facing problems with Unicall on this scenario :

CentOS 4.3 - Running on x86_64
Asterisk 1.2.7.1
Zaptel 1.2.5

When running zttool , shows all Spans OK.

But I can't receive and make calls.

I tried to change many parameters and still doesn't work.

Any clues ?

* unicall.conf 

[channels]

language=br

context=incoming-pstn
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancellforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes

rxgain=0.0
txgain=0.0
faxdetect=both
loglevel=255
protocolclass=mfcr2
protocolvariant=br,20,4
protocolend=cpe
group=1
callgroup=1

channel => 1-15
channel => 17-31
channel => 32-46
channel => 48-62
channel => 63-77
channel => 94-108
channel => 110-124

* zaptel.conf *

loadzone=br
defaultzone=br


span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101

span=2,0,0,cas,hdb3
cas=32-46:1101
cas=48-62:1101


span=3,0,0,cas,hdb3
cas=63-77:1101
cas=79-93:1101

span=4,0,0,cas,hdb3
cas=94-108:1101
cas=110-124:1101



* lor error *

-- Executing Dial("SIP/1000-1de2", "Unicall/g1/40020022|40|Ttr") in new
stack
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Call control(1)
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Make call
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Making a new call with CRN 32769
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 0001  ->  [1/   1/Idle  /Idle ]
-- Called g1/40020022
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
Unicall/1 event Dialing
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1  <-   [1/  40/Seize /Idle ]
Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 4 on  ->  [2/  40/Group I   /Idle ]
Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 R2 prot. err. [2/  40/Group I   /DNIS ] cause
32769 - T1 timed out
Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 4 off ->  [1/   1/Idle  /Idle ]
Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event:
Unicall/1 event Protocol failure
-- Unicall/1 protocol error. Cause 32769
Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Channel echo cancel
Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Channel gains
Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 Channel switching
-- Hungup 'UniCall/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/1000-1de2' status is 'CHANUNAVAIL'
Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1  <- 1011  [1/   1/Idle  /Idle ]
Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/1 1001  ->  [1/   1/Idle  /Idle ]

Jefferson Carvalho

 


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[Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?

[Shaun]

End result was i gave in for a annoying setup... put the phone and a server 
on a network by it self, setup a dhcp server with a tftp address and flashed 
the phone to 8.2 which also reset the password back to cisco

-- 

~Shaun

"Shaun" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
> Anybody know the proceedure to factory reset the a 7960 phone running 6.3 
> SIP software?  I've tried holding # when booting the phone and nothing, i 
> can do that on my 8.2 phone but this phone i just got with 6.3 isnt 
> working. Also **# doesnt work either..
>
> -- 
>
> ~Shaun
>
>
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