[asterisk-users] allo.com gsm card with AsteriskNOW

[Nicolas Pham Van Huyen]

>Dear all,
>
>I'm searching someone who already installed allo.com gsm card 
>with AsteriskNOW.
>
>I installed the hardware in my new server, but when running dahdi_genconf 
>always have the "no span" message.
>I I tried to install the driver according to allo.com doc, but it seems not up 
>to
>date, and maybe done more specifically for an appliance..
>
>>[root@localhost ~]# dahdi_genconf
>>Empty configuration -- no spans
>>Empty configuration -- no spans
>>Empty configuration -- no spans
>
>Thanks for help
>Nico

 
David Duffett wrote :
>Step 1 would be an 'lspci' on the Linux command line to see if the Linux box 
>recognises the card Step 2 would be to ensure that your DAHDI version is new 
>enough to work with the card

Yes , the card is present :
>04:00.0 Bridge: PLX Technology, Inc. Device d44e (rev 01)

Jg wrote :
>Don't they have a kernel module that communicates with the card on one and 
>with DAHDI 
>on the other side? The first steps are probably to check with lspci whether 
>the card is 
>detected and then make sure the allo module is loaded.

They have a package 
http://www.allo.com/firmware/gsm-card/chan_allogsm-1.1.2_P2.tar.gz

with an installer inside (install.sh), but it is turned as full installer :
- download and install  dahdi-linux-complete-2.5.0+2.5.0.tar.gz
- seems makes some compilation for the driver 
- finishes by proposing to install asterisk 

As I installed AsteriskNOW before, it makes no sense to install Asterisk again, 
I just need to install the driver

In the package I can found src folders, but I don't know how to use them to 
build the module manually.
The installer seems building allog4c.ko file but it doesn't works installed by 
this script

I guess it is just a basic operation to know how to compile the module and load 
it every time the system boots
Then how to configure the channels " chan_allogsm" ..

Thanks for help :-)
Nico




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[asterisk-users] allo.com gsm card with AsteriskNOW

[Nicolas Pham Van Huyen]

Dear all,

I'm searching someone who already installed allo.com gsm card with Asterisk NOW.

I installed the hardware in my new server, but when runing dahdi_genconf I 
always have the "no span" message.
I tried to install the driver according to allo.com doc, but it seems not up to 
date, and maybe done more specificly for an appliance..

>[root@localhost ~]# dahdi_genconf
>Empty configuration -- no spans
>Empty configuration -- no spans
>Empty configuration -- no spans

Thanks for help
Nico

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[asterisk-users] Ringing detection ?

[Nicolas Ross]

Hi !

I have an application where I originate a call with a call file and play 
some pre-recorded message when the person answers. And it's working 
correctly.


Now, I've been asked to add the support for extenstion numbers.

I've been able to actualy send the extension numbver via the SendDTMF 
command. It works perfectly. But after the dtmf have been sent, the 
dialplan shoud wait for someone to answer.


The problem is that during that time, the phone is rining. So withing 
north-america, it's typicly ring 2 secs, silence 4 seconds.


So, I could use AMD() WaitForSilence(4100) for exemple. But that would 
require the person at the other end to be silent for 4 seconds. That's 
unrealistic.


So, I'm searching for a way for my dialplan to detect ringing and only 
lauch Amd/WaitforSilence after the person answers...


My curent ael dialplan for my originated call is :

  500 => {
Answer();
Wait (1);
if (${LEN(${noPoste})})
{
  SendDTMF(${noPoste});
}
Background(silence/1);
AMD();
WaitForSilence(500);
for (x=0; ${x} < 3; x=${x} + 1) {
  Background(outcall/outcall-${idJob});
  Background(outcall-confirm);
  WaitExten(5);
};
goto diffuseurappel|3|1;
  };

Any ideas ?

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Re: [asterisk-users] Problem with libpri / asterisk

[Nicolas Ross]

(...)

My guess is your new setup is trying to do a PRI 2B Transfer (meaning 
that Asterisk is trying to handoff two B channels of a PRI to the 
upstream switch).  It is probably being rejected and the call is hanging 
up.  You will need to dig into the PRI debug of both scenarios and 
compare.   I was not even aware that Asterisk could do that so it may be 
some new feature being worked on.

I just found this:  http://wiki.sangoma.com/Asterisk-FAQ#TBCT

Maybe you should check and see if it is enabled.


My god !

That was it !

It was enabled, I had transfer = yes, but there was no mention of 
facilityenable. I disabled it, restarted asterisk, and voilà !


Thanks for pointing that out ! 



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[asterisk-users] Problem with libpri / asterisk

[Nicolas Ross]

Hi all !

We currently have an asterisk box that is rather old (runs Asterisk
1.4.21.2), and it's connected to the PSTN with a sangoma A104d card.

Now we have a new PRI at another location, and I use that occasion to
build 2 new servers, one to replace our aging one and a new one for this new
pri.

So I downloaded the lastest libpri / asterisk / wanpipe driver, but the
previous version of dahdi (2.5), since the latest wanpipe isn't compatible
with dahdi 2.6. All is built from source

Now, all seems to be working OK. I can connect a SIP phone to my new box,
make calls to the outside, receive calls etc.

But, I can't seem to bridge a call. So on my new server, with the new PRI, I
got a Sangoma a104 card (no echo-canceler on this one).

In my extensions.ael, I got this :

418nx1 => {
 Answer();
 Wait (2);
 Playback(demo-thanks);
 Dial(${TRUNK}/418nx2);
};

TRUNK is DAHDI/G1

Where 418nx1 is a DID on my new PRI and 418nx2 is my cellphone
number.

When I do a call from my home phone or cell phone to my new PRI to
418nx1, I hear the demo-thanks file, and then it dials out. My cellphone
rings, but as soon as I pick up the call, the calls hangs up :

   -- Accepting call from '418nx2' to '418nx1' on channel 0/1, span 
1
   -- Executing [418nx1@ael-default:1] Answer("DAHDI/i1/418nx2-b", 
"") in new stack
   -- Executing [418nx1@ael-default:2] Wait("DAHDI/i1/418nx2-b", 
"2") in new stack
   -- Executing [418nx1@ael-default:3] 
Playback("DAHDI/i1/418nx2-b", "demo-thanks") in new stack

   --  Playing 'demo-thanks.ulaw' (language 'fr')
   -- Executing [418nx1@ael-default:4] Dial("DAHDI/i1/418nx2-b", 
"DAHDI/G1/418nx2") in new stack

   -- Requested transfer capability: 0x00 - SPEECH
   -- Called DAHDI/G1/418nx2
   -- DAHDI/i1/418nx2-c is proceeding passing it to 
DAHDI/i1/418nx2-b

   -- DAHDI/i1/418nx2-c is ringing
   -- DAHDI/i1/418nx2-c is making progress passing it to 
DAHDI/i1/418nx2-b

   -- DAHDI/i1/418nx2-c answered DAHDI/i1/418nx2-b
   -- Native bridging DAHDI/i1/418nx2-b and DAHDI/i1/418nx2-c
   -- Span 1: Channel 0/1 got hangup request, cause 16
   -- Hungup 'DAHDI/i1/418nx2-c'
 == Spawn extension (ael-default, 418nx1, 4) exited non-zero on 
'DAHDI/i1/418nx2-b'

   -- Hungup 'DAHDI/i1/418nx2-b'

BUT, if I originate the call from my curent PRI, it goes in and out and all
is well. I noticed that if the calls go trough correctly and hangup 
manually, it also stats the exact same thing (cause 16). So the above 
console output might not be that much usefull...


I've had a case open with Sangoma for this issue, and they suggested I go 
the libpri/asterisk for more help debuging this issue, since on their end, 
the disconnect comes from the telco...


They suggested I try a different version of asterisk, wich I did to no 
avail, or try there NBE product instead of libpri...


So, did anybody ever encontered something like that ? What steps should I 
take to diagnose the problem furhter ?


Thanks for any help. 



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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Show us the output of a failed call with pri debug enabled on that span.
It will be difficult, since the PRI is in use on our "old" asterisk box. 
I will have to get to the colo at night, to avoid disrupting calls 
during the day.


Is there any other thing that I should collect ?

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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

In asterisk CLI do "pri show spans". The fact the card is in RED alert
means the hardware does not "see" the pri line connected to the card.


I probably made a mistake in copying / pasting. pri show spans was showing 
something like :


PRI span 1/0: Provisioned, Up, Active

Calls can enter, I see them arriving on the console, but they imediatly got 
hangun, cause 6. 



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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Le 2011-05-09 09:31, Jim Dickenson a écrit :

Make sure the firmware on the card is latest. I had a problem, not like your, 
and flashing the card to the latest firmware resolved it.

It appears it did not change anything...

So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, 
on the asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1.


When asterisk is running, cat /proc/dahdi/1 yields :

Span 1: WPT1/0 "wanpipe1 card 0" (MASTER) B8ZS/ESF RED

   1 WPT1/0/1 Clear (In use)
   2 WPT1/0/2 Clear (In use)
(...)
  24 WPT1/0/24 Hardware-assisted HDLC (In use)

And when it's not, the (In use) go away.

When, dialing I get "Unable to create channel of type 'DAHDI' (cause 34 
- Circuit/channel congestion)"


So, does anybody got any idea ?

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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Make sure the firmware on the card is latest. I had a problem, not like
your, and flashing the card to the latest firmware resolved it.


I did the upgrade, I will make another test when appropriate.

I will also upgrade my curent card, I am curent at version 25, wich dates < 
2007, it might solve our curent problem also...


Regards,



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[asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Hi !

We curently have a centos 5 / asterisk 1.4 server that we have some DTMF 
problems with. It has a Sangoma A104d card and only port one is used to 
connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for 
modem access and port 3 is connected for data communication via PPP.


Now, I want to freshen this setup to something newer. So I installed a 
Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers 
and an A101 card I had laying around.


I did a test this weekend and pluged in our PRI in that test server. I never 
got succeded to have a call trough. When I dialed in, the call is "hanged 
up" with :


Channel 1/1, span 1 got hanup, cause 6
Spawn extension (ael-default, s, 3) exited non-zero on 
'DAHDI/i1/NPANXX-2'

Hungup 'DAHDI/i1/NPANXX-2'

Here's my dahdi/system.conf :

loadzone=us
defaultzone=us

#Sangoma A101 port 1 [slot:0 bus:6 span:1] 
span=1,1,0,esf,b8zs
bchan=1-23
echocanceller=mg2,1-23
hardhdlc=24

my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with :

switchtype=national
pridialplan=unknown
signalling=pri_cpe
group=1
channel => 1-23

as the last non-commented lines.

So, for one thing, the card I have in my test server doesn't have an 
hardware echo canceller, but it's still enabled in my wanpip setting. Could 
that be a source of problem ?


Other than that, is there anything obvious I've missed ? 



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Re: [asterisk-users] Under heavy attack

[Nicolas Ross]

I just wanted to add my voice to this "attack". I saw the morning that I had 
200+ distinct ips since the weekend. I used a small perl script that blocks 
failed usernames and passwords at iptables level I found thei morning :

http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block/

Regards,
  My main asterisk server is under unusual heavy attack, and so far Fail2Ban 
has blocked about 30 IPs, from various different countries. At this time it is 
blocking about 1 IP address every few minutes.
  Just wondering if anybody else is also experiencing unusually increased hack 
attempts today?


  Zeeshan A Zakaria
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Re: [asterisk-users] Pickup parcked call from Aastra9480i ctcordless

[Nicolas Ross]

> Is the phone defined as a SIP extension/peer?  If so, try "sip set debug
> peer xxx" and try the call/pickup again.

Yes, and doing so, the phone could no longer dial out, bizare.

>> Yes, after I can pick it up from my phone (9133i), and it works. I had
>> verbosity at 6 at the moment of testing. When he enters 701, only his
>> phones
>> displays "Failed", nothing in asterisk. I can pickup after that on mine.
>
> Is there a dial plan on the phone that you need to alter?

Yes : "x+#|xx+*", same as mine.

After the problem with dial-out stated on top, we rebooted the phone another 
time, and now everithing works... That was strange... 


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Re: [asterisk-users] Pickup parcked call from Aastra 9480i ctcordless

[Nicolas Ross]

> Make sure your verbosity is set to at least 5 and try to see the CLI 
> output
> on failure again.  Are you sure the call is parked on 701 (not 702-720 as
> defined in features.conf)?

Yes, after I can pick it up from my phone (9133i), and it works. I had 
verbosity at 6 at the moment of testing. When he enters 701, only his phones 
displays "Failed", nothing in asterisk. I can pickup after that on mine.

> Other calls/extension dials work from this phone?

Yes, our extensions are 3 digits, and all other calls to / from this 
(problematic) phone works.

Regads, 


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[asterisk-users] Pickup parcked call from Aastra 9480i ct cordless

[Nicolas Ross]

Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park 
call feature of asterisk to transfer calls to one another.

But the 9480i ct cordless cannot pickup a parked call. When manually 
entering 701 (parked call extention), the phone display "Call failed" (appel 
écoué in french).

Nothing is displayed on the asterisk console.

When doing it from other aastra phones (same config), or other make phones, 
it works.

Any hints on possible causes ? 


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[asterisk-users] [SIP/H.264] Codec negotiation problem ?

[Nicolas Bourbaki]

Hi,

I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : "every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated"

What I observe :
  - a call made from a SIP Phone registred in Asterisk to Tandberg works
(voice and video bidirectionnal)
  - a call made from to Tandberg to the SIP phone doesn't work (voice
bidirectionnal, voice only received by the SIP phone, no incomming video for
Tandberg)

I think the problem may  come from codec negotiotation :
  - when call is made from the SIP phone, it uses "code" 99 for H.264 codec,
as Asterisk. Tdb reply SIP:Ok with the same number for H.264
  - when call is made from Tbd, it uses "code" 98 for H.264 codec. Asterisk
then send the Invite with 99 as codec number


I use the version 1.6.2.6 of Asterisk

Is this kind of configuration supposed to work ? I know passing video media
through Asterisk may not be optimal, but I really need it, even if I have to
patch Asterisk

Thanks for your help



SDP send by Tandberg :
--
v=0
o=tandberg 1 5 IN IP4 192.168.50.10
s=-
c=IN IP4 192.168.50.10
b=CT:1920
t=0 0
m=audio 48260 RTP/AVP 100 101 9 8 0 102
b=TIAS:64000
a=rtpmap:100 G7221/16000
a=fmtp:100 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-15
a=sendrecv
m=video 48262 RTP/AVP 97 98 99 34 31
c=IN IP4 192.168.50.10
b=TIAS:192
a=rtpmap:97 H264-RCDO/9
a=fmtp:97 profile-level-id=008016;max-
mbps=42000;max-fs=3600;max-smbps=323500
a=rtpmap:98 H264/9
a=fmtp:98
profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500
a=rtpmap:99 H263-1998/9
a=fmtp:99
custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo
a=rtpmap:34 H263/9
a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200
a=rtpmap:31 H261/9
a=fmtp:31 cif=1;qcif=1;maxbr=19200
a=rtcp-fb:* nack pli
a=sendrecv
a=content:main
a=label:11
a=answer:full
m=application 5078 UDP/BFCP *
c=IN IP4 192.168.50.10
a=floorctrl:c-s
a=confid:1
a=floorid:2 mstrm:12
a=userid:1
a=setup:passive
a=connection:new
m=video 48264 RTP/AVP 99 34 31
c=IN IP4 192.168.50.10
b=TIAS:192
a=rtpmap:99 H263-1998/9
a=fmtp:99
custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo
a=rtpmap:34 H263/9
a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200
a=rtpmap:31 H261/9
a=fmtp:31 cif=1;qcif=1;maxbr=19200
a=rtcp-fb:* nack pli
a=sendrecv
a=content:slides
a=label:12



SDP send by Asterisk
v=0
o=root 1077353049 1077353049 IN IP4 192.168.13.100
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.13.100
b=CT:384
t=0 0
m=audio 14604 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17962 RTP/AVP 99
a=rtpmap:99 H264/9
a=sendrecv





Here is my sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
disallow = all
realm = testRealm

allow = ulaw
allow = h264
videosupport=yes

canreinvite = no
calleridupdate = info
usercallerid = no
context = default

[toTandberg]
host=192.168.50.53
type=friend
qualify=yes
qualifyreq=1
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[asterisk-users] Redone setup, bizare problems

[Nicolas Ross]

Hi !

Sorry if this is a long post...

I had this setup for about a year without problems :

Network A   <->  wrv200  <->  internet  <->  wrv200  <-> net b

The 2 networks are linked with an ipsec vpn. The 2 internet connections are 
with the same cable company to minimize latency, both separates /24 subnets.

On network A, I got 2 computers, a single sip phone (aastra 9112i). On 
network B, I got a linux box acting as smb domain controler and many other 
thins, another linux box that only do an asterisk server, 4 computers, 3 sip 
phones (2 aastra 9112i and one 480i ct), 1 analog ATA for the alarm system, 
an internet point of sale terminal, that's about it.

The asterisk server is connected to the PSTN with a VOIP provider trough 
IAX.

All of this was working fine. Since the last couple of months, we had some 
qos problems with the exterior. The sound was choping up randomly. With what 
I can do with the linksys wrv200, I was limited. I also had some filesystem 
issues that made the asterisk server locks up from time to time. So I wanted 
to reformat the server and re-install it on a ssd drive.

So for the time of the re-format, I installed asterisk on my smb server, and 
re-copied /etc/asterisk, /var/spool/asterisk and /var/lib/asterisk to it.

When I arrived at home, the externel power supply of the little shuttle box 
refused to power up. So I ended up building a new computer from scratch. 
This new server will be acting as asterisk server and router, replacing the 
wrv200 from network b (demoting it to wireless access point and switch).

Last sunday, I installed the server, re-copying the 3 folders of asterisk in 
the same maner. I then had a hard time making the phones register to the 
server. I always had "no service" with the mwi light steady on. I finaly got 
the phones to register, I'm still not sure exactly what I did to make it 
work. And the phone on network A didn't work wither. I'll get to that one 
later. The linksys pap2t ata didn't had the problem, it registered right as 
I started the server.

On the phones on network B, since then I get the MWI light come on 
momentarely, with the "no service" on the display, and then all comes back 
to normal. But all phones can make and receive calls.

For the remote phone, for those familiar with ipsec vpns, it's a net to net 
connection. So, the gateway cannot reach the computers on the network on the 
other side. So I had to add a static route on my router/asterisk server to 
be able to reach the phone on the other side. It was able to register for 
some time, but the next morning it was "no service", and I wasn't able to 
make it work again. I ended up connecting the phone trough the externel 
interface of the router/asterisk server.


So, my questions :

Why is the phones are constatly showing "no service" with the light flashing 
?

Is there a way for the remote phone from net A to connect properly trough 
the vpn ?


Thanks for any hints,

Niolas 


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[asterisk-users] RLT Transfers and NI2

[Nicolas Chapleau]

Hello Users,

 

   quick question for all you fine PRI users out there using RLT transfers. 
Scenario:

 

1) Inbound call received on channel one

2) Call is transferred to an agent on channel 2

3) RLT is enabled on this PRI, thus channels are released on the Asterisk box

 

So far, all is fine, but our PRIs have 4 enabled RLT channels, thus after the 
5th transfer, while the previous calls are still active, we get a busy 
(congestion), when we in fact have many PRI channels left. Again, this is a 
normal opration for RLT. My question to you is: Can we attempt a Bridged 
transfer instead of a RLT upon reception of a Congestion RLT result? 

 

Specifics:

 

Asterisk 1.4.24.1

libpri 1.4.9

Allstream DS3 muxed out to 28 PRIs, b8zs, esf, NI2 signalling

 

Hope this makes sense to one of you.

 

Nic.
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Re: [asterisk-users] ATA for large networks

[Nicolas Ross]

I personnlay found that marc is better than google when searching mailing 
lists :

http://marc.info/?l=asterisk-users&r=1&w=2

> What is the best-recommended resource for searching archives of this 
> mailing
> list?
>
> Thanks for your time


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[asterisk-users] Transfer a call without announce : no sound

[Nicolas Ross]

When we receive a call from outside (via a sangoma 104d card) and we do a
"blind transfer", that is without anouncing to the called party , we have no
sound either way.

Exemple :

I take my cell phone to call my * box, it rings on my aastra 9113i phone, I
answer.
Then I hit the xfer buton, make my second call to another extention (it can
be either a aastra phone, nortel phone trough ciel portico, whatever.
As soon it rings I hangup or hit the xfer buton again.
Then the bridged call between the other extension and the zap channel have
no sound either way.

If I wait for the called party to answer and announce the transfer, all is
fine.

I've had report of sound one way also, but I wasn't able to reproduce.

Here's the log from my console :

   -- SIP/224-09e0f098 answered Zap/1-1
-- Started music on hold, class 'default', on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/224-09e1d728",
"ael-std-exten|225|SIP/225") in new stack
-- Executing [EMAIL PROTECTED]:1] Set("SIP/224-09e1d728",
"ext=225") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/224-09e1d728",
"dev=SIP/225") in new stack
-- Executing [EMAIL PROTECTED]:3] Answer("SIP/224-09e1d728", "") in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp("SIP/224-09e1d728",
""Nicolas Ross" <224>") in new stack
-- Executing [EMAIL PROTECTED]:5] Wait("SIP/224-09e1d728", "0.5")
in new stack
-- Executing [EMAIL PROTECTED]:6] Dial("SIP/224-09e1d728",
"SIP/225|15") in new stack
-- Called 225
-- SIP/225-09e73388 is ringing
-- Stopped music on hold on Zap/1-1
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728' in macro 'ael-std-exten'
  == Spawn extension (macro-ael-std-exten, s, 6) exited non-zero on
'SIP/224-09e1d728'
-- SIP/225-09e73388 answered Zap/1-1

Any ideas ? 


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Re: [asterisk-users] Interfacing pri card to legacy pbx

[Nicolas Ross]

I cannot tell for sure for any system, but we have an old Portmaster PM3 
hooked-up from one port of our Sangoma A104d card, another one being from 
telco.

So, yes you can "emulate" the telco from a sangoma A10x card. Here's what I 
have in my zapata.conf :

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

;Sangoma A104 port 2 [slot:12 bus:0 span: 2]
echocancel=no
pridialplan=national
signalling=pri_net
group=2
channel => 25-47

You might have noticed that the signalling is different for both port. 
pri_net being the "telco emulatin" one. The clock needs also to be set on 
"master" in the wancfg utility.

Another thing, you might want to consider using a 2 port card for that, 
because the clock master needs a reference and I can't tell for sure if 
it'll work with a reference from another card.

Regards,

Nicolas


> Hi guys,
> Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server?
> The pbx doesn't have sip and I want to come in off of a sip trunk and
> interface with the older system.
> I know I can use a pri card to hook in to the phone network, but can I use
> this same card to feed back the signaling as if I were the phone company 
> to
> the older system?
>
> Thanks,
> Tom


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Re: [asterisk-users] Asterisk on SLOW solid state disk

[Nicolas Ross]

Hi !

We are running our asterisk from a transcend ts8gifd25. The whole system, 
including the OS fit in this 8 gig disk. If you don't do any recording of 
calls, you don't need that much of speed.

We have 20 or so SIP phones, a PRI trough a quad-port sangoma card, one other 
port is a "pass-trough" to a dial-up RAS, another port is a point-to-point T1 
data link to our office.

To date, we've handle a handfull of simultaniously calls without any 
performance degradation.

Regards,

I'm looking at building up a standard asterisk system fanless/no moving 
parts.  I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it is 
SLOW...25mb/sec read 8mb/sec write.



Has anyone tried a slow disk like this on asterisk?  Will this delay voice 
prompts or screw up ast/linux in any interesting way?



(I know there are linux distros and Asterisk projects designed to run off 
CF, but I'm hoping to stay mainstream)



Thanks,
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[asterisk-users] SIP Phone, multiple line, but one call at a time ?

[Nicolas Ross]

Hi !

I got some users how have SIP hard phones (aastra for one and nortel trough 
a Citel Portico TVA for the other), who doesn't want the phone to ring when 
they are on the phone. I still need for the "line" to be displayed at least 
2 times on the phone for them to make transfer and conferences, so I just 
cannot make the sip acount only availaible on one button...

Is there any variable that could tell me the "status" (i.e. InUse) of the 
phone that I could use in my dialplan ?

Any other ideas ?

Thanks,

Nicolas 


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[asterisk-users] chan_sip Maximum retries exceeded on transmission

[Nicolas Ross]

I have a situation here where a user has an AAstra 480i phone, which 
function corectly. The phone is behing a nat-router (a linksys wrv200 for 
it's VPN point to point facility). The phone is plugued in a port wich has 
qos enabled.

And when the user places a call, sometimes (not always), we get this in the 
console :

[May  8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 1867350652 
(Critical Response)
[May  8 13:41:55] WARNING[5804]: chan_sip.c:1972 retrans_pkt: Hanging up 
call [EMAIL PROTECTED] - no reply to our 
critical packet.

And then, the call stops.

I did enabled sip debug for that peer, and I saw that a packet was sent to 
the phone, and the phone did reply (I see it in the debug).

In sip.conf, for that peer, I have nat=yes.

Any ideas ? 


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Re: [asterisk-users] SMS gateway recommendation

[Nicolas Ross]

Here, we've tested in the past zimsms, but now the,re closed. We founded out 
that the most reliable and cost-effective way to send sms is with GPRS 
modem. Multitech manufacures excelent quality product for gprs, edge and 
others.

So we opened up an sms-only acount, it costs us 10$ / month for 1000 sms 
sent, plus 10c each after, plus some other fees like 911 fees, etc. We 
generelay sent no more than a couple hundeds.

With this modem hookup-up to a serial port, and a little program assuring 
the sending / receiving of sms (sms_link from Sourceforge)., and voilà !

Here in Canada, all cellular providers are hooked-up one to another for 
passing of sms', so I send out my sms' with Rogers, but any user in any 
network receives it in about 1 to  5 seconds. I don't know if it's the same 
in the US or other countries.

If you have more questions about the specifics, you can contact me.

Nicolas

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

Sent: Monday, December 24, 2007 6:54 AM
Subject: Re: [asterisk-users] SMS gateway recommendation


>> Hash: SHA1
>>
>>
>> I personally use clickatell.
>
> Can you please comment on the reliability of clickatell, pricing and
> speed of delivery of the message to the end-user?
>
>> But I use a PHP script to do the texting and call that from the dialplan.
>
> Is that script available somewhere?
>
> Thanks,
> Patrick


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

[Nicolas Ross]

We've finaly solved it...

For once, when we plugued the PRI into the first port, all began to work 
better.

As for my dial-out into my RAS, I had to put pridialplan=unknown into my 
channels group of incoming line, and pridialplan=national in my group for my 
pri going to my RAS. That did the trick.

Thanks,

Nicolas

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

Sent: Thursday, November 01, 2007 10:31 AM
Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster)


> Thanks,
>
> That's not it.
>
> I juste uninstalled wanpipe, redownloaded zaptel (another version 
> (1.4.5.1)
> to try), re-installed wanpipe, (patching zaptel in the process), recompile
> wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything,
> and still got the same errors...
>
> Nicolas
>
>
> - Original Message - 
> From: "Jared Smith" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Thursday, November 01, 2007 9:15 AM
> Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster)
>
>
>> On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote:
>>> I also get sometime :
>>>
>>>   == Primary D-Channel on span 2 down
>>> [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
>>> D-channels available!  Using Primary channel 48 as D-channel anyway!
>>>   == Primary D-Channel on span 2 up
>>
>> I don't claim to be an expert on the Wanpipe drivers, but it's been my
>> experience that if your D-channels bounce up and down like that every
>> few seconds, that the Wanpipe drivers didn't successfully patch Zaptel,
>> or you haven't restarted Zaptel since the Wanpipe drivers patched
>> Zaptel.
>>
>> Hopefully that's enough to get you up and running.


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

[Nicolas Ross]

Thanks,

That's not it.

I juste uninstalled wanpipe, redownloaded zaptel (another version (1.4.5.1) 
to try), re-installed wanpipe, (patching zaptel in the process), recompile 
wanpipe, re-compiled zaptel, recompiled asterisk, re-installed everything, 
and still got the same errors...

Nicolas


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

Sent: Thursday, November 01, 2007 9:15 AM
Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster)


> On Wed, 2007-10-31 at 20:57 -0400, Nicolas Ross wrote:
>> I also get sometime :
>>
>>   == Primary D-Channel on span 2 down
>> [Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
>> D-channels available!  Using Primary channel 48 as D-channel anyway!
>>   == Primary D-Channel on span 2 up
>
> I don't claim to be an expert on the Wanpipe drivers, but it's been my
> experience that if your D-channels bounce up and down like that every
> few seconds, that the Wanpipe drivers didn't successfully patch Zaptel,
> or you haven't restarted Zaptel since the Wanpipe drivers patched
> Zaptel.
>
> Hopefully that's enough to get you up and running.
>
>
> -- 
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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> 


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

[Nicolas Ross]

Here's my planed setup :

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

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

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

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

Zaptel.conf:

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

zapata.conf (part of) :

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

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



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

On * console, I get :

Primary D-Channel on span 2 up

All the time

I also get sometime :

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


In my extensions.conf, I have :

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

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

Any help would be appriciated.

Nicolas 


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Re: [asterisk-users] Jajah.com like script?

[Nicolas Brisac]

you can also have a look at 
http://www.voip-info.org/wiki/index.php?page=Asterisk+click+to+call


I implemented the PHP script mentioned in the comments and it works like 
a charm.
It will require some modification to achieve what you want to do, but 
that shouldn't be too difficult.




Steve Edwards wrote:

Or supply the second channel as the data to the dial application.

Ritesh, please read the documentation and experiment a bit so you can 
ask more challenging questions :)


On Fri, 16 Mar 2007, mitcheloc wrote:


You are missing something. Initiate the call to channel for the first
user (i.e. ZAP/g1/phonenumber), and have their destination extension
the second phone number.

On 3/16/07, Ritesh Agrawal <[EMAIL PROTECTED]> wrote:

So does this mean that we have to dump the two callers into a Meetme
context???
The problem there is what if one of the callee's doesn't accept the 
call

(call screening).
There is no easy way to kick the other user out of meetme and dump 
him to a

vmail context.
Am I missing something?

R


On 3/16/07, Ritesh Agrawal <[EMAIL PROTECTED]> wrote:
> Thanks Steve!
> I will give it a shot.
>
> R
>
>
>
> On 3/16/07, Steve Edwards < [EMAIL PROTECTED]> wrote:
> > Search on voip-info.org for call files.
> >
> > On Fri, 16 Mar 2007, Ritesh Agrawal wrote:
> >
> > > Hi Folks,
> > >
> > > I am planning to create an internal portal where the users can 
enter

two
> > > phone numbers (theirs and the party they are trying to reach) and
connect
> > > the two of them by initiating two calls from Asterisk. Any 
pointers on

how
> > > to initiate two calls and then bridge them (without using 
meetme?).

> > > Ideally, I would like to do a call screening as well.
> > >
> > > Thanks for your help.
> > >
> > > R
> > >
> >
> > Thanks in advance,
> >
 

> > Steve Edwards  [EMAIL PROTECTED]   Voice: 
+1-760-468-3867

PST
> > Newline
Fax: +1-760-731-3000
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> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>


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--

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com
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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline     Fax: +1-760-731-3000
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--
Nicolas Brisac

Solutions First
9476 0076
0404 849 629
http://www.solutionsfirst.com.au

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[asterisk-users] extensions.conf gotoif and label

[Nicolas]

Hello,

I got a little interogation about these 3 points.
I want to write something like this sample in my extension.conf. I have 
tested and it works but I don't know if it is a good way to make a menu.

I don't want to put number as it is boring to maintain.
Does anyone know if there is some problem to write like this?

exten => 7890,1,Wait(1)
exten => 7890,n(lbl0),Read(REP|annonce|1)
exten => 7890,n,GotoIf($[${REP} = 1 ] ?lbl1:lbl2)
exten => 7890,n(lbl1),noop( hit 1 ! )
exten => 7890,n,system(echo you hit one)
exten => 7890,n,Hangup
exten => 7890,n(lbl2),GotoIf($[${REP} != 2 ] ?lbl3)
exten => 7890,n,noop( hit 2 ! )
exten => 7890,n,system(echo you hit two)
exten => 7890,n,Hangup
exten => 7890,n(lbl3),noop( hit something else ! )
exten => 7890,n,system(echo you hit another key)
exten => 7890,n,goto(lbl0)

Another question on the web i have seen there is some trouble using 
"while" application. Is that still true or it was an old release wich 
get this problem?



Thanks ,
Nicolas

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Re: [asterisk-users] How to know who hangup ?

[Nicolas]

Gregory Duchatelet a écrit :


Hi,

Using AMI or dial plan, how can i know which leg (channel ?) of a 
bridged call, hangup ?


AMI send 2 hangup events, which have both cause 16 (normal clearing), 
and the first hangup event is the called leg hangup event, not the one 
who hangup…


Greg



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adding g in your dial application and the call will go on the extension 
when the callee hangup

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Re: [asterisk-users] Monitoring awareness

[Nicolas]

I think you are right or i didn't find how to to it without using a 
conference.

And even with conference didn't find a smart way to make it.

Ondrej Valousek a écrit :

Hi Steve,

Ok Playback could be used here, indeed.
But if you are using automonitor - by default activated by (*1) - I 
think there is no way how to implement this.

Am I right?

Thanks,
Ondrej

Steve Totaro wrote:

[EMAIL PROTECTED] wrote:

Hello,

I'm discovering asterisk, it seem to be a great soft.

I have seen a fonction to record calls that's a great fontion but 
there is

something disturbing me.

When the record start, except if the recorder prevent the other 
part, he is not

aware of the recording...

I dont find a way from the feature.conf how to play a sound when a 
monitor start

to record :/

  
Either play a file with a beep or a verbal message that this call may 
be recorded for such and such reason.  This can be done easily in the 
dialplan by calling playback or background prior to monitor.


Depending on local laws, you may be OK if just one party on the call 
knows it is being recorded.  Other states have different laws.  I 
have no idea how the law works when one caller is in one state with 
one set of laws and the other caller is in a different state with 
different laws.


Thanks,
Steve
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Nicolas S.]

I didnt forgot the french translation, it's coming was just busy. 
In 1-2 weeks i ll provide it to you 

Regards


Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :
> Hi all!
> 
> We've released VoiceOne 0.4.0, a web-based and open source solution 
> which allows to fully manage an Asterisk service hosted on a LAMP server.
> 
> We focused on an charming and overall user-friendly interface. Thanks to 
> the authentication based on roles, once configured by a super user, the 
> PBX may be easily maintained even by an Asterisk unskilled users.
> 
>  From a technical point of view, the application is made up of two 
> modules: one for the client - i.e. the user interface - and the other 
> for the server. Thanks to the web services provided by the server module 
> and the use of a database, VoiceOne may be easily integrated with other 
> applications (e.g. CRM software).
> 
> The project has grown and has received positive response so far. 
> Nowadays there's a little but enthusiastic community of developers, 
> supporters and users. Translations in several languages (e.g. English, 
> Spanish, Russian, etc.) are already available.
> 
> On the project website at http://www.voiceone.it you'll find the online 
> demo and the links to download the source files from Sourceforge, as 
> well as a support forum.
> 
> We would be pleased if you could give it a try and let us know your 
> feedback, comments, ideas, or suggestions replying here or posting a 
> message on our forum.
> 
> Thanks for your kind attention.
> 
> Regards,
> Alex
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[asterisk-users] Feature and multiple application

[Nicolas]

Hello,

I am making some experiments with asterisk.
I dont find a way to make a simple application: During a bridged call, i 
pressed several touch (ie: #234#) what i want to start is a message then 
another saying the number the caller dialed.


Step 1: Caller join the callee, they discuss
Step 2: The caller forgot the number he dialed and he wished to know it
Step 3: dial #234#
Step 4: playback(thenumberyoudialed)
Step 5: saydigit{${exten})

With the file features.conf, i cant start only 1 application.

Can someone give me a way to find how to make it?

Thanks for reply,
Nicolas
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Nicolas S.]

Hi, 

I can help in French translation if needed.
Drop me the procedure to do it. 

Regards 


Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :
> Hi all!
> 
> We've released VoiceOne 0.4.0, a web-based and open source solution 
> which allows to fully manage an Asterisk service hosted on a LAMP server.
> 
> We focused on an charming and overall user-friendly interface. Thanks to 
> the authentication based on roles, once configured by a super user, the 
> PBX may be easily maintained even by an Asterisk unskilled users.
> 
>  From a technical point of view, the application is made up of two 
> modules: one for the client - i.e. the user interface - and the other 
> for the server. Thanks to the web services provided by the server module 
> and the use of a database, VoiceOne may be easily integrated with other 
> applications (e.g. CRM software).
> 
> The project has grown and has received positive response so far. 
> Nowadays there's a little but enthusiastic community of developers, 
> supporters and users. Translations in several languages (e.g. English, 
> Spanish, Russian, etc.) are already available.
> 
> On the project website at http://www.voiceone.it you'll find the online 
> demo and the links to download the source files from Sourceforge, as 
> well as a support forum.
> 
> We would be pleased if you could give it a try and let us know your 
> feedback, comments, ideas, or suggestions replying here or posting a 
> message on our forum.
> 
> Thanks for your kind attention.
> 
> Regards,
> Alex
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[asterisk-users] Asterisk MGCP Centrex Experience/success story

[Nicolas S.]

Hello, 

I have recently a practical with students about Asterisk and MGCP, I m
used to SIP and asterisk concepts, but I d like to know if someone has
some experience or success story about Centrex features with MGCP (as
far I know it s the most used protocol for centrex). 
I' d like to show the students some Centrex features ( I dont know much
of it ). And how a Centrex can differ from a standard PBX 
It s not really a technical help I need but more an approach to show
Asterisk can do Centrex too. 
Any experience would be nice. 

Kind regards. 

Nicolas S. 

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Re: [asterisk-users] cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons

[Nicolas S.]

Hi, 

Can you paste us the error messages when you compile cdr_addon_mysql

Best regards 


Le jeudi 12 octobre 2006 à 03:07 +0100, Marco Mouta a écrit :
> Hi guys,
> 
> I've been installing Asterisk 1.4 with Asterisk addons, and i could
> notice that in /usr/lib/asterisk/modules/ doesn't have
> cdr_addon_mysql.so even after compiling Asterisk Addons!
> 
> In fact the cdr_addon_mysql.c exists, but it doesn't seems to be
> compile when i run Asterisk-Addons: make && make install 
> 
> Any one can help me on this?
> 
> -- 
> best regards
> 
> Marco Mouta 
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Re: [asterisk-users] asterisk - alcatel

[Nicolas Bocquet]

Hello, We have test this configuration but we think it's a problem with the Alcatel.how are you doing to make the trunk between alcatel and Asterisk?We use a card PRA recommended by an Alcatel's technician and you?
ThanksNicolasOn 9/26/06, Frederico Madeira <[EMAIL PROTECTED]> wrote:
I'm trying the same in Alcatel 4200 and  i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira
[EMAIL PROTECTED]
2006/9/26, Sylvain ZUCCA <[EMAIL PROTECTED]>:

Hi,
 
can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX
 
Best Regards. 
2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>:
Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: 
http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). 
But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___

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-- Sylvain 

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[Asterisk-Users] Fax to Email issue with Spandsp tif not correctly sized

[Nicolas FINETIN]

Hello, I've google to trouble shoot my issue but i was not able to find a solution, I've install asterisk all the libraries to receive faxes:Spandsp 0.0.2 pre 25 tiff lib.I'make it work i can receive faxes on an extension but the issue is that i m in France an we are sending faxes in A4 format so i set that up in the perl script /var/lib/asterisk/bin/fax-
process.pl by passing the good option to the tiff2pdf.still not good output then i check the .tif   and here i could see that the tif itself is not well formatted.It is looking like larger is bigger and height is smaller.
How can i set the rx_fax.c that i well receiving the Fax's tif.Thank you Regards,-- -= Nicolas Finetin =-[EMAIL PROTECTED]
+33 689 20 90 72
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[Asterisk-Users] extension.conf for overlap

[Nicolas LEGROS]

Hi !!

 

I’d like to confgure my « extensions.conf »
file in order to handle overlap!!

Is it possible?

What should be changed?

 

Thanks by advance

Nicolas L.






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Re: [Asterisk-Users] CARD.XML for MGCP cisco phone

[Nicolas TOUSSAINT]

Anyone has a working CARD.XML for cisco
MGCP phones?
 The one on the cisco site is old
and it's not working with the new 
 firmwares.
 
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[Asterisk-Users] Asterisk & MGCP reinvite

[Nicolas TOUSSAINT]

Hi all, 
I'm having troubles with the MGCP canreinvite option. 
I have two eyep media and Asterisk on the same network, same codecs, the
option line "canreinvite=yes" properly set in the mgcp.conf file...
but it won't do. 
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Re: [Asterisk-Users] Swissvoice IP10S centralized phonebook

[Nicolas Olivier]

This is the information I got from Swissvoice support, I didn't tried yet, but 
if it can helps.


How to use an external phone book

IP10S phone supports access to Cisco Phone Book but not all functionalities. 
The IP10 uses his own interface to access to the Phone Book.
If you want to connect to your remote phone book, you have to do the following 
actions:

First, copy the URL under Search by name in a Web browser, for example:
http://192.168.1.5/cisco/directory/searchDirectory.php

You are going to have a XML file display in the Web Browser, like this one:

  Directory Search
  Enter search criteria
  http://10.3.100.190:8080/ciscodirectory?action=list&page=0
  
First Name
firstname
A
  
  
Last Name
lastname
A
  
  
Number
number
T
  


Copy the information from the URL line 
(http://10.3.100.190:8080/ciscodirectory?action=list&page=0).

The easiest way to set the path in your phone is by the Web interface (but it 
could also be done by Telnet).
Connect to your phone web server. Login and password are normally: admin
Select Configure common phonebook.
In Select phone book to use chose the value: Remote
Then click on submit.
File the box below with the URL you get previously:
The IP address and the port number of the Phonebook server can be manually 
entered or synchronised with the Call Agent
In our case, IP address: 10.3.100.190, Port number: 8080,Path: 
/ciscodirectory?action=list&page=0

Then click on submit. If you return to your phone and select the common phone 
book, it is normally connected to the remote one now.
You can search by a name or if you put nothing and press on OK, it will return 
you the entire content of the remote phone book.

More information about Cisco Phonebook management
To manage remote phone book, you need a server. It could be one you developed by yourself or one include in your proxy server (not all of them include 
this feature). It must follow the Cisco implementation; you can have more information here:

http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/3_1/index.htm
Check for "Cisco IP Phone Services Application Development Notes with Cisco 
CallManager 3.1."


Igor Briski wrote:


Anybody got any documentation/experience on the subject?

I'm trying to get it working, but the documentation I have lacks any
information on what should be installed on the server side.

--
Igor Briški - [EMAIL PROTECTED]
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[Asterisk-Users] SIP gateway: call hangups afer 3 rings

[Nicolas Olivier]

Hi all,

I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another 
asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are 
not answered, and from the debug trace, it seems to be the asterisk gw
who hangups. Apart from that, calls answered before three rings are handled 
correctly.
I don't really see what could explain such comportement, and can't find a 
"related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a 
call.

Best regards,
Nicolas Olivier


The gateway is running asterisk 1.0.7.

sip.conf:

[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes

[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes

extensions.conf:

[default]
exten => _x.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _x.,2,Hangup
exten => _x.,3,Congestion

(...)

Call debug:

-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2, 
actual format = 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "SIP/[EMAIL PROTECTED]") in 
new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to zzz.zzz.zzz.zzz:5060
-- Called [EMAIL PROTECTED]


Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact: 
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: 
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact: 
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: ;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201

v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20

11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc 
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 
(nothing)
-- SIP/b3g-7bfa is making progress passing it to IAX2/[EMAIL PROTECTED]/1
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
  == Spawn extension (default, 0123456789, 4) exited non-zero on 'IAX2/[EMAIL 
PROTECTED]/1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'


Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: "Choco Bobo" ;tag=as1a492e28
To: 
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0

(...)


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[Asterisk-Users] SIP gateway: call hangups afer 3 rings

[Nicolas Olivier]

Hi all,

I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for 
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are not answered, and from the debug trace, it seems to be the asterisk gw 
who hangups. Apart from that, calls answered before three rings are handled correctly.

I don't really see what could explain such comportement, and can't find a 
"related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a 
call.

Best regards,
Nicolas Olivier


The gateway is running asterisk 1.0.7.

sip.conf:

[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes

[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes

extensions.conf:

[default]
exten => _x.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _x.,2,Hangup
exten => _x.,3,Congestion

(...)

Call debug:

-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2, 
actual format = 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "SIP/[EMAIL PROTECTED]") in 
new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to zzz.zzz.zzz.zzz:5060
-- Called [EMAIL PROTECTED]


Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact: 
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: 
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact: 
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: ;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201

v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20

11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc 
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 
(nothing)
-- SIP/b3g-7bfa is making progress passing it to IAX2/[EMAIL PROTECTED]/1
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
  == Spawn extension (default, 0123456789, 4) exited non-zero on 'IAX2/[EMAIL 
PROTECTED]/1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'


Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: "Choco Bobo" ;tag=as1a492e28
To: 
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0

(...)

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[Asterisk-Users] Yuxin hardphones feedback

[Nicolas Schmerber]

Hello everybody

This question has probably already been asked, but I'd like to have 
feedbacks about Yuxin hardphones
Especially series 10, 100 and 200 ( and by the way i didnt found too 
much technical differences between those models). Is it better than 
budgettone, or so cheap hardphones ?
I m'  like everybody and dont want to waste money in bad phones :) So 
any experience is welcome.



Thanks in advance




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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

The VoIP Connection a écrit :


Nicolas,

Just did some quick testing and the instructions are incorrect.  You need to
press "transfer" to complete the transfer instead of the second "flash".
This actually makes more sense.

Attended and regular transfer both work perfectly with the following
settings:

Enable Call Features: "Yes"
Disable call Waiting: "No"
Send Flash event: "No"

DTMF should be whatever * is set to, but in-band won't work properly if your
codec is anything other than U-Law.

By the way, the newest firmware also makes the long overdue conference
feature work properly.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]


 


-Original Message-
From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] 
Sent: Thursday, August 11, 2005 10:41 AM
To: [EMAIL PROTECTED]; Asterisk Users 
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Supervised transfer problem 
with BudgetTone


The VoIP Connection a écrit :

   


Section 4.3.7.2 from the Bugetone Manual:

The user can transfer an active call to a third party with 
 


announcement.
   

The user presses the “flash” button and hears a dial tone, then dial 
the 3rd party’s phone number followed by pressing send 
 

button. If the 
   

call is answered, press “flash” to complete the transfer 
 

operation, if 
   

the call is not answered, pressing “flash” button to resume the 
original call.


Notes:

• If attended Transfer fails, the BudgeTone phone will ring 
 

the user to 
   

remind that another party is still on the call, the user can 
 

then pick 
   


up the call using handset or speaker.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]




 


-Original Message-
From: Nicolas Schmerber [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 5:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Supervised transfer problem with 
BudgetTone


[EMAIL PROTECTED] a écrit :

  

   


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:





 

All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but
  

   


none gave
  

   


me the correct answer (unless I missed it).
 

  

   


Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in


 


order to
  

   


have supervised transfers.

Steve

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When looking at a recent firmware changelog of Grandstream 
   

, it says 
   


BT should support supervised transfer, so shouldnt it work ?


  

   


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Tried this manipulation a few minutes ago :

A calls B , B pushes flash button ( A is waiting with a mp3 
played) B calls C pressing Send ; C answers B presses flash 
button again ; C is so on hold (with a mp3 played) B hangs up 
But A and C arent in connect ; the phoneof B rings ( to tell 
someone is in wait : A)


So it seems to fail

What should i put in grandstream config for the next item :
/Enable Call Features: Y/ N ?
//Disable Call-Waiting: Y/N ?
//Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO 
/Send Flash Event: Y / N ? / Any others Ideas ?.


Thx

Nicolas S.

   



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Thank you all, now it works
The last method (grandsteam manual but with transfer key instead) was 
the right


Thanks

Nicolas



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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

The VoIP Connection a écrit :


Section 4.3.7.2 from the Bugetone Manual:

The user can transfer an active call to a third party with announcement.
The user presses the “flash” button and hears a dial tone, then dial the 3rd
party’s phone number followed by pressing send button. If the call is
answered, press “flash” to complete the transfer operation, if the call is
not
answered, pressing “flash” button to resume the original call.

Notes:

• If attended Transfer fails, the BudgeTone phone will ring the user to
remind that
another party is still on the call, the user can then pick up the call using
handset
or speaker.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]


 


-Original Message-
From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] 
Sent: Thursday, August 11, 2005 5:59 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Supervised transfer problem 
with BudgetTone


[EMAIL PROTECTED] a écrit :

   


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:



 

All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but 
   

none gave 
   


me the correct answer (unless I missed it).
  

   


Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in 
 

order to 
   


have supervised transfers.

Steve

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When looking at a recent firmware changelog of Grandstream , 
it says BT should support supervised transfer, so shouldnt it work ?



   



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Tried this manipulation a few minutes ago :

A calls B , B pushes flash button ( A is waiting with a mp3 played)
B calls C pressing Send ;
C answers
B presses flash button again ;
C is so on hold (with a mp3 played)
B hangs up
But A and C arent in connect ; the phoneof B rings ( to tell someone is 
in wait : A)


So it seems to fail

What should i put in grandstream config for the next item :
/Enable Call Features: Y/ N ?
//Disable Call-Waiting: Y/N ?
//Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO
/Send Flash Event: Y / N ? /
Any others Ideas ?.

Thx

Nicolas S.
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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

Eric Wieling aka ManxPower a écrit :


Olle E. Johansson wrote:


CVS head of Asterisk supports attended transfers native in Asterisk, not
really SIP attended transfers. Work is in progress in that area, but
will require quite a lot of changes to the SIP channel so I am not sure
whether we will be able to support it in 1.2 or not. Definitely in the
1.4 release.



What is the specific problem? We hav been doing supervised transfers 
with 1.0.x and Polycom phones for several months.



Thanks for answering me all, but seems it s a debate to see if it works :)
I m not able to have other phones for the moment, so if this kind of 
transfer doesnt work with Budgetones it doesn't matter, but if someone 
had successfull story with it , I would apreciate much.


Any other idea maybe ?

Nicolas S.
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Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

[EMAIL PROTECTED] a écrit :


On Thu, 11 Aug 2005, Nicolas Schmerber wrote:

 

All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but none gave me 
the correct answer (unless I missed it).
   




Hi,

You'll need to switch to the CVS-HEAD version of Asterisk in order to have 
supervised transfers.


Steve

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When looking at a recent firmware changelog of Grandstream , it says BT 
should support supervised transfer, so shouldnt it work ?

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[Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

Hi all,

I'm quite new on this mailing list, and I discover the asterisk world.
I m experimenting a PBX with SIP phones, grandstream budgetone (not 
expensive for tests)


All the features I need work just not one : the supervised call 
transfers. I know there are a lot of posts about that, but none gave me 
the correct answer (unless I missed it).


Here is my config :

2 sip phones BT102 with firmware : 1.0.6.7 (the last at this day)
Asterisk from debian stable (1.0.7)
a Bri connection arives on the PBX, this works , and can reach the sip 
phones; but the call transfer only works in blind ( no ability to speak 
to the transfee to introduce the incoming call).

Here are config files :

extensions.conf :

NICO = SIP/nico
CEDRIC = SIP/cedric

[default]
include => incoming
exten => 22,1,Dial(${CEDRIC},20)
exten => 23,2,Dial(${NICO},20)

[incoming] ; the BRI stuff
exten => 9692,1,Dial(${CEDRIC},20)   ; if numerber arriving on bri 
finishes by 9692 dial Cedric

exten => _969X,1,Dial(${NICO},20)  ; else dial Nico


features. conf :

[general]
atxfer => *5

So I d like to know the params for the BT phones, the asterisk config , 
and the procedure ( for example should i press *5 when i want to release 
the line and etablish caller => transfee ) and so on .


Thanks in advance







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[Asterisk-Users] CISCO 7960 with Asterisk

[Nicolas Boittin]

Hi,

 

We are trying to set up
an asterisk configuration using some 7960 Cisco Telephone. We need to deploy
those in our company and we also need to see on the screen who is on line or
not. After making a research on the web, we thing that we have to use MGCP or
sccp.

 

Does anybody have the
last firmware of Cisco 7960 to work either in SCCP or MGCP?

 

Rgds,

 

Nicolas






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RE: [Asterisk-Users] VoiceMail

[Nicolas FOURNIL]

Check your voicemail.conf !! you can do your own date prompt (or no date
prompt)


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Waldo
Rubinstein
Envoye : lundi 20 juin 2005 15:56
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] VoiceMail


I installed Asterisk Voicemail in an office and now most of the
employees are complaining that when they're listening to the
messages, it takes "forever" to listen to their messages. The reason
being is that before the message is played, the voicemail says the
full date and time when the message arrived and that takes a long
time. It's like: Friday . June20th.
2000...and...5... etc (you get the idea). Is there anyway
to shorten that or even give users the option to not play that?

Thanks,
Waldo
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Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Stuart,

I switched the system to a pentium based host, with different memory.
The results are the same. I've also changed the ISDN card to be sure.

Nicolas

Stuart Hirst wrote:
> Nicolas,
> 
> I replied earlier stating that I saw similar issues and now that you
> have applied the Florz patch the symptoms you are seeing are all but
> identical to the issues I saw and resolved by changing out the
> motherboard memory. The system was an ASUS main board with a Xeon
> processor.
> 
> It is not the memory it could be something specific to the VIA motherboard.
> 
> Stuart
> 
> 
> 
> Nicolas Olivier wrote:
> 
>>Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded
> after ztcfg with:
>>
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> underrun: 0, 0
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> overflow: 311, 311
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> overflow: 436, 436
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> underrun: 0, 0
>>
>>And when I start asterisk, same stuff, kernel crashes.
>>
>>Interrupts are ok.
>>
>>sjaak imap wrote:
>> 
>>
>>>Dear Nicolas Olivier
>>>
>>>Just try the florz patch at http://zaphfc.florz.dyndns.org/
>>>and look at cat /proc/interupts if your not sharing irq's
>>>
>>>Maybe this will help
>>>
>>>
>>>Good luck
>>>
>>>Sjaak
>>>
>>>   
>>>
>>>>Hi,
>>>>
>>>>I'm trying to setup a small BRI ISDN <-> voip gateway.
>>>>The ISDN card is based on Cologne chipset, so I try set it up with
> zaphfc.
>>>>
>>>>The versions i'm running:
>>>>kernel-2.4.27
>>>>Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>>>>zaptel modules 1.0.7
>>>>zaphfc is from bristuff-0.2.0-RC8e
>>>>
>>>>When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>>>
>>>>Zapata Telephony Interface Registered on major 196
>>>>PCI: Found IRQ 12 for device 00:12.0
>>>>zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
>>>> 
>>>>
>>>0xc2d58000(0x2d58000) IRQ 12 HZ 100
>>>   
>>>
>>>>zaphfc: Card 0 configured for TE mode
>>>>Registered Span 1 ('ZTHFC1') with 3 channels
>>>>Span ('ZTHFC1') is new master
>>>>zaphfc: 1 hfc-pci card(s) in this box.
>>>>Registered Span 2 ('ZTRTC/1') with 0 channels
>>>>Real Time Clock Driver v1.10e
>>>>
>>>>I'm using zaprtc as the gateway is running on a VIA motherboard without
>>>> 
>>>>
>>>USB controller.
>>>   
>>>
>>>>When I'm doing ztcfg -vv:
>>>>
>>>>Zaptel Configuration
>>>>==
>>>>
>>>>SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>>>>
>>>>Channel map:
>>>>
>>>>Channel 01: Individual Clear channel (Default) (Slaves: 01)
>>>>Channel 02: Individual Clear channel (Default) (Slaves: 02)
>>>>Channel 03: D-channel (Default) (Slaves: 03)
>>>>
>>>>3 channels configured.
>>>>
>>>>Here are my confs:
>>>>
>>>>/etc/zaptel.conf:
>>>>
>>>>loadzone=fr
>>>>defaultzone=fr
>>>>
>>>>span=1,1,3,ccs,ami
>>>>bchan=1-2
>>>>dchan=3
>>>>
>>>>/etc/asterisk/zapata.conf:
>>>>
>>>>[channels]
>>>>
>>>>language=fr
>>>>context=test
>>>>switchtype=euroisdn
>>>>signalling=bri_cpe
>>>>echocancel=yes
>>>>immediate=yes
>>>>channel => 1-2
>>>>
>>>>/etc/asterisk/modules.conf:
>>>>
>>>>[modules]
>>>>autoload=yes
>>>>
>>>>noload => pbx_gtkconsole.so
>>>>noload => pbx_kdeconsole.so
>>>>
>>>>noload => app_intercom.so
>>>>
>>>>load => chan_modem.so
>>>>load => res_features.so
>>>>load => res_musiconhold.so
>>>>load => chan_zap.so
>>>>
>>>>noload => chan_alsa.so
>>>>noload => chan_oss.so
>>>>
>>>>[global]
>>>>chan_modem.so=yes
>>>>

Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by 
ztdummy, zaprtc or zaprai.

But anyway the results are the same with or without zaprtc loaded.

Peer Oliver Schmidt wrote:
> Nicolas Olivier wrote:
> 
>> I'm trying to setup a small BRI ISDN <-> voip gateway.
>> The ISDN card is based on Cologne chipset, so I try set it up with
> zaphfc.
>>
>> The versions i'm running:
>> kernel-2.4.27
>> Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>> zaptel modules 1.0.7
>> zaphfc is from bristuff-0.2.0-RC8e
>>
>> When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>
>> Zapata Telephony Interface Registered on major 196
>> PCI: Found IRQ 12 for device 00:12.0
>> zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
>> zaphfc: Card 0 configured for TE mode
>> Registered Span 1 ('ZTHFC1') with 3 channels
>> Span ('ZTHFC1') is new master
>> zaphfc: 1 hfc-pci card(s) in this box.
>> Registered Span 2 ('ZTRTC/1') with 0 channels
>> Real Time Clock Driver v1.10e
>>
>> I'm using zaprtc as the gateway is running on a VIA motherboard
> without USB controller.
> [..]
> 
> Why are you running zaprtc? zaphfc provides your needed timing source.
> -- 
> Best regards
> 
> Peer Oliver Schmidt
> PGP Key ID: 0x83E1C2EA
> 
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Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded after 
ztcfg with:

May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
311, 311
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
436, 436
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0

And when I start asterisk, same stuff, kernel crashes.

Interrupts are ok.

sjaak imap wrote:
> Dear Nicolas Olivier
> 
> Just try the florz patch at http://zaphfc.florz.dyndns.org/
> and look at cat /proc/interupts if your not sharing irq's
> 
> Maybe this will help
> 
> 
> Good luck
> 
> Sjaak
> 
>>Hi,
>>
>>I'm trying to setup a small BRI ISDN <-> voip gateway.
>>The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
>>
>>The versions i'm running:
>>kernel-2.4.27
>>Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>>zaptel modules 1.0.7
>>zaphfc is from bristuff-0.2.0-RC8e
>>
>>When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>
>>Zapata Telephony Interface Registered on major 196
>>PCI: Found IRQ 12 for device 00:12.0
>>zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
>>zaphfc: Card 0 configured for TE mode
>>Registered Span 1 ('ZTHFC1') with 3 channels
>>Span ('ZTHFC1') is new master
>>zaphfc: 1 hfc-pci card(s) in this box.
>>Registered Span 2 ('ZTRTC/1') with 0 channels
>>Real Time Clock Driver v1.10e
>>
>>I'm using zaprtc as the gateway is running on a VIA motherboard without
> USB controller.
>>
>>When I'm doing ztcfg -vv:
>>
>>Zaptel Configuration
>>==
>>
>>SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>>
>>Channel map:
>>
>>Channel 01: Individual Clear channel (Default) (Slaves: 01)
>>Channel 02: Individual Clear channel (Default) (Slaves: 02)
>>Channel 03: D-channel (Default) (Slaves: 03)
>>
>>3 channels configured.
>>
>>Here are my confs:
>>
>>/etc/zaptel.conf:
>>
>>loadzone=fr
>>defaultzone=fr
>>
>>span=1,1,3,ccs,ami
>>bchan=1-2
>>dchan=3
>>
>>/etc/asterisk/zapata.conf:
>>
>>[channels]
>>
>>language=fr
>>context=test
>>switchtype=euroisdn
>>signalling=bri_cpe
>>echocancel=yes
>>immediate=yes
>>channel => 1-2
>>
>>/etc/asterisk/modules.conf:
>>
>>[modules]
>>autoload=yes
>>
>>noload => pbx_gtkconsole.so
>>noload => pbx_kdeconsole.so
>>
>>noload => app_intercom.so
>>
>>load => chan_modem.so
>>load => res_features.so
>>load => res_musiconhold.so
>>load => chan_zap.so
>>
>>noload => chan_alsa.so
>>noload => chan_oss.so
>>
>>[global]
>>chan_modem.so=yes
>>chan_zap.so=yes
>>
>>
>>The problem is that after ztcfg ran, I've got the following logs:
>>
>>Registered tone zone 2 (France)
>>zaphfc: card 0 layer 1 state = F4
>>zaphfc: card 0 layer 1 state = F5
>>zaphfc: card 0 layer 1 state = F7
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 layer 1 state = F3
>>zaphfc: card 0 layer 1 state = F4
>>zaphfc: card 0 layer 1 state = F5
>>zaphfc: card 0 layer 1 state = F7
>>zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
> wanted 8 got 7), probably a buffer overrun.
>>zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
> wanted 8 got 7), probably a buffer overrun.
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>
>>And when I start asterisk -c, same logs keep on, and I've finally a
> kernel crash:
>>
>>Unable to handle kernel paging request at virtual address fffc
>> printing eip:
>> c0113cc0
>> *pde = d063
>> *pte = 
>> Oops: 
>> CPU:0
>> EIP:0010:[]Not tainted
>> EFLAGS: 00010013
>> eax: c248015c   ebx:    ecx: 0001   edx: 0001
>> esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
>> ds: 0018   es: 0018   ss: 0018
>> Process sshd (pid: 146, stackpage=c2c8f000)
>> Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
> c3

Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Just an update, I deoopsed the kernel dump, must be usable...

Nicolas Olivier wrote:
> 
> Hi,
> 
> I'm trying to setup a small BRI ISDN <-> voip gateway.
> The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
> 
> The versions i'm running:
> kernel-2.4.27
> Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
> zaptel modules 1.0.7
> zaphfc is from bristuff-0.2.0-RC8e
> 
> When I'm doing the insmod on zaptel, zaphfc, zaprtc:
> 
> Zapata Telephony Interface Registered on major 196
> PCI: Found IRQ 12 for device 00:12.0
> zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
> zaphfc: Card 0 configured for TE mode
> Registered Span 1 ('ZTHFC1') with 3 channels
> Span ('ZTHFC1') is new master
> zaphfc: 1 hfc-pci card(s) in this box.
> Registered Span 2 ('ZTRTC/1') with 0 channels
> Real Time Clock Driver v1.10e
> 
> I'm using zaprtc as the gateway is running on a VIA motherboard without
> USB controller.
> 
> When I'm doing ztcfg -vv:
> 
> Zaptel Configuration
> ==
> 
> SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
> 
> Channel map:
> 
> Channel 01: Individual Clear channel (Default) (Slaves: 01)
> Channel 02: Individual Clear channel (Default) (Slaves: 02)
> Channel 03: D-channel (Default) (Slaves: 03)
> 
> 3 channels configured.
> 
> Here are my confs:
> 
> /etc/zaptel.conf:
> 
> loadzone=fr
> defaultzone=fr
> 
> span=1,1,3,ccs,ami
> bchan=1-2
> dchan=3
> 
> /etc/asterisk/zapata.conf:
> 
> [channels]
> 
> language=fr
> context=test
> switchtype=euroisdn
> signalling=bri_cpe
> echocancel=yes
> immediate=yes
> channel => 1-2
> 
> /etc/asterisk/modules.conf:
> 
> [modules]
> autoload=yes
> 
> noload => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> 
> noload => app_intercom.so
> 
> load => chan_modem.so
> load => res_features.so
> load => res_musiconhold.so
> load => chan_zap.so
> 
> noload => chan_alsa.so
> noload => chan_oss.so
> 
> [global]
> chan_modem.so=yes
> chan_zap.so=yes
> 
> 
> The problem is that after ztcfg ran, I've got the following logs:
> 
> Registered tone zone 2 (France)
> zaphfc: card 0 layer 1 state = F4
> zaphfc: card 0 layer 1 state = F5
> zaphfc: card 0 layer 1 state = F7
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 layer 1 state = F3
> zaphfc: card 0 layer 1 state = F4
> zaphfc: card 0 layer 1 state = F5
> zaphfc: card 0 layer 1 state = F7
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
> wanted 8 got 7), probably a buffer overrun.
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
> wanted 8 got 7), probably a buffer overrun.
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> 
> And when I start asterisk -c, same logs keep on, and I've finally a
> kernel crash:
> 
> Unable to handle kernel paging request at virtual address fffc
>  printing eip:
>  c0113cc0
>  *pde = d063
>  *pte = 
>  Oops: 
>  CPU:0
>  EIP:0010:[]Not tainted
>  EFLAGS: 00010013
>  eax: c248015c   ebx:    ecx: 0001   edx: 0001
>  esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
>  ds: 0018   es: 0018   ss: 0018
>  Process sshd (pid: 146, stackpage=c2c8f000)
>  Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
> c3819545
> 0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
> 0086
> c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
> c270c800
> Call Trace:[] [] [] []
> []
>   [] [] [] [] []
> []
>   []
> 

>>EIP; c0113cc0 <__wake_up+20/a0>   <=

>>eax; c248015c <_end+217f0d0/34fef74>
>>esi; c24803a0 <_end+217f314/34fef74>
>>edi; c248015c <_end+217f0d0/34fef74>
>>ebp; c2c8fe2c <_end+298eda0/34fef74>
>>esp; c2c8fe14 <_end+298ed88/34fef74>

Trace; c3819545 <[zaptel]__zt_receive_chunk+133d/1484>
Trace; c01cb6b1 <__ide_do_rw_disk+3e1/650>
Trace; c381aae6 <[zaptel]zt_receive+a26/b0c>
Trace; c381aad7 <[zaptel]zt_receive+a17/b0c>
Trace; c383cd78 <[zaphfc]hfc_interrupt+228/358>
Trace; c01cae16 
Trace; c383ce95 <[zaphfc]hfc_interrupt+345/358>
Trace; c01c5416 
Trace; c01cad01 
Trace; c0109ddd 
Trace; c0109f78 
Trace; c010c328 

Code;  c0113cc0 <__wake_up+20/a0>

[Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

038
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
(...)
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) 
ITU-2 VoiceModem Driver)
  == Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
 [res_features.so] => (Call Parking Resource)
  == Parsing '/etc/asterisk/features.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '700' priority 1 to parkedcalls
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
  == Registered application 'HoldedCall'
  == Registered application 'AutoanswerLogin'
  == Registered application 'Autoanswer'
 [res_musiconhold.so] => (Music On Hold Resource)
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
 [chan_zap.so]

The ISDN line has been validated, and the ISDN is known to work. I've searched 
in the archives, wiki, and can't see what's wrong.
If anyone has an advice, it will be greatly appreciated.

Nicolas Olivier


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RE: [Asterisk-Users] French SIP or IAX phones

[Nicolas FOURNIL]

Videotel !!! : French software, Video hard phone, Excellent browser...

see it at : http://www.call.fr

Works fine with Asterisk.



-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Martin Roy
Envoye : vendredi 13 mai 2005 00:52
A : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] French SIP or IAX phones


Is there any SIP or IAX phones that can be configure in french  
instead of english. I tested Cisco 7960 phones but I can't change the  
language it's only available in english with the SIP firmware.

I have a customer that's located in France and he wants french phones  
if possible. So I'm wondering if there's any one out there that found  
a phone that can be change to french.

Thanks

Martin Roy
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RE: [Asterisk-Users] beginner in Asterisk configuration

[Nicolas FOURNIL]

Hello

Sorry for english speaking peaple, but I just help this beginner in our
natural language : French ;-)

Je suis Français aussi, si tu as besoin d'un peu d'aide tu peux me joindre
directement par mail

Pour tester ta config : asterisk -gc

Bonne chance

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Tutu Lord
Envoyé : jeudi 12 mai 2005 09:58
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] beginner in Asterisk configuration


hello,

i am french student and i want configure a Asterisk server.

when I want launch the server with the command safe_asterisk -vcf

the server answer : Asterisk ended with exit status 1
   Asterisk died with code 1

what is the signification of it please ?

thank you

lucas

_
MSN Hotmail : antivirus et antispam gratuits
http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR

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RE: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!

[Nicolas FOURNIL]

Same problems as you...

Eyebeam is not really fine in video... We have find some nasty bugs in it
(PC freeze, codecs issues...) and no feedback from Xten after sending back
reports (tcpdump and long descriptions). I think that EyeBeam works fine
with... eyebeam. The software seems to be beta because of each version of
Eyebeam I've download has differents bugs.

Try with our hard-videophone ( ;-) ), Asterisk video features works. Perhaps
a small problem in Intra Frame request (I've posted it in feature request
without success). We will work on it ASAP.

Nicolas
http://www.call.fr


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Shadow
Roldan
Envoye : mardi 1 mars 2005 20:40
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!


Hi All

First time poster, long time reader.

I just ran into something really bizarre. I've enabled videosupport and
have been testing sip calls with Xten Eyebeam software, it works
(mostly)

However, when I have

Videosupport=yes

In the [general] section of my sip.conf, broadvoice calls fail w/ "We're
sorry your call cannot be completed at this time"


So... I've commented it out and tried adding videosupport=yes to
specific extensions, now video doesn't work as eyebeam reports "remote
user does not support video" but broadvoice works.

Bizarre

I'm running CVS v1-0-02/15/05

Any ideas?


_

Shadow Roldan
IT Manager
Zero G Software, Inc.

tel: +1.415.512.7771 x 306
fax: +1.415.723.7244
mailto:[EMAIL PROTECTED]
www.ZeroG.com

The leading provider of multiplatform software deployment solutions.
_

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Re: [Asterisk-Users] ATA's

[Nicolas Bougues]

On Tue, Feb 15, 2005 at 01:40:24PM -0600, Matthew Boehm wrote:
> > wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is
> > a "bridge mode", where the LAN and WAN ports would act just like a
> > switch, so that you can easily chain devices without routing/NAT. Just
> > like most IP phones do.
> 
> So what happens if you try and chain a bunch of SPA-2100s? Does the 2100
> act more like a router?
> 

The SPA-2100 will perform NAT between the LAN and WAN ports. It means
that you won't be able to connect to devices "behind" SPA-2100 from
the WAN side (except by configuring DMZ, but it's gets awful then).

It also means that you can't chain them with their basic config,
because it won't like having the same network address (192.168.1.x) on
both its interfaces.

So right now, if you want to chain them, you have to play with IP
addresses, DHCP settings, etc. Not fun, particularly when you consider
that this device is really "plug and play", it remotes configure
everything.

Hopefully a "bridge" mode will appear in a later firmware upgrade
(which, for Sipuras, are frequent and readily available on their
website).

-- 
Nicolas Bougues

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Re: [Asterisk-Users] ATA's

[Nicolas Bougues]

On Mon, Feb 14, 2005 at 10:47:23PM +0900, Hermann Wecke wrote:
> Matthew Boehm wrote:
> >[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
> >has both WAN/LAN ports.
> 
> I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying 
> one to test. Street price around US$ 90.
> Another one with dual g729 channels is MTA V102. Street price US$ 100. 
> Also will test this one.
> 
> I'm still looking for other units with dual g729 channels...
> 

Back in december, the Uniden "was supposed to do 2xG729 at a later
time". Not sure if the current firmware allows it.

BTW, I've been fairly disappointed with Uniden firmware and their
release cycle : their hardware is great, but they take months to
release new firmwares, even when "phone crashing" bugs are
discovered.

If you want 2xG.729 now, working reliably, for under $90, you can't go
wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is
a "bridge mode", where the LAN and WAN ports would act just like a
switch, so that you can easily chain devices without routing/NAT. Just
like most IP phones do.

-- 
Nicolas Bougues

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Re: [Asterisk-Users] ATA's

[Nicolas Bougues]

On Sun, Feb 13, 2005 at 10:39:36AM -0800, Luki wrote:
> The Sipuras have a ton of configurable parameters. If you understand
> them (and there is no good manual, unfortunately) then you can be of
> great benefit. Otherwise they'll be worthless. I particularly miss the
> dial-plan, distinctive ring and audio gain options on the
> Grandstreams. Remote syslog can also be useful for debugging. It all
> depends what you need, I guess.
> 
> Further, the Sipuras have a more detailed status, that is accessible
> WHILE you are engaged in a conversation.
> 
> I think you're paying a bit more for the 1000 (1 line version) as
> compared to the Grandstream 286, but if you need/want two independent
> lines, then the Spa 2000 is more economical (as Peter said).
> 

The Sipuras are really a dream to manage, particularly in an
international environment. You can customize the tones, the rings, the
voltages, the dialplan, the features... well, everything.

They are (securely) remote manageable and upgradeable. They are rock
solid. Sipura support is helpful in case you need them for complex
issues. Voice quality is top notch.

The Grandstreams are less manageable, have less parameters, have only
american tones, no dialplan support, no auto-upgrade (well, they
recently added some kind of support). Voice quality is OK.

-- 
Nicolas Bougues

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Re: [Asterisk-Users] ATA's

[Nicolas Bougues]

On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote:
> We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN
> port. Only downside is that only 1 call can be using 729 at a time. This has
> been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to
> overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and
> has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100.
> 

My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2
for the SPA-2000.

-- 
Nicolas Bougues
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Re: [Asterisk-Users] Asterisk and Sipura SPA-841 SIP phones

[Nicolas Bougues]

On Thu, Feb 10, 2005 at 09:07:58AM -0500, Giovanni Powell wrote:
> Nothing to do with your question, but by any chance, when you plugged
> the phone into the wall did you hear a dialtone or is this something
> generated by asterisk

On a SIP phone, the dial tone is locally generated. The Sipura will
only generate a dial tone if registrered. BTW, you can easily check on
the Sipura web interface that the dial tones are parametered there.

-- 
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] high-quality, high-bandwidth codecs?

[Nicolas Bougues]

On Tue, Feb 08, 2005 at 02:58:01PM +0100, Roy Sigurd Karlsbakk wrote:
> >>are there any codecs around that allows high quality as in "studio
> >>lite"? it may consume high bandwidth, and hopefully allow some packet
> >>loss.
> >>
> >
> >I'm not sure what "studio lite" means to you. Maybe hard figures would
> >be more precise.
> >
> >G.722 might be interesting : 64 kbps, 7 kHz. It's not free.
> >
> >Otherwise, MP3 or OGG might be ok ?
> 
> Would it be hard to do a codec_ogg?
> 

It would rather be a codec_vorbis, as Steve pointed out. It's
definetly feasible. However, I'm not sure how useful it would
be. You'd need some kind of device talking Vorbis to Asterisk. Does it
exist ?

-- 
Nicolas Bougues

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Re: [Asterisk-Users] high-quality, high-bandwidth codecs?

[Nicolas Bougues]

On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote:
> hi
> 
> are there any codecs around that allows high quality as in "studio 
> lite"? it may consume high bandwidth, and hopefully allow some packet 
> loss.
> 

I'm not sure what "studio lite" means to you. Maybe hard figures would
be more precise.

G.722 might be interesting : 64 kbps, 7 kHz. It's not free.

Otherwise, MP3 or OGG might be ok ?

-- 
Nicolas Bougues

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Re: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade

[Nicolas Chabbey]

Thanks for your reply.
I've already seen the wiki page concerning the xml configuration file
problem and mine have absolutly no comments on it, just the minimal
lines needed.
I've just tried to put the 'P0S30202' on OS79XX and as expected it
crash. I think i'll not be able to upgrade to SIP by this way as all
name begining with P0S3 or P0M (for MGCP) will automatically hang the
device.




On Thu, 03 Feb 2005 09:26:56 -0500, Steve Blair <[EMAIL PROTECTED]> wrote:
> 
> Nicholas:
> 
>   You need to "convert" from SCCP to SIP by loading  image_version: P0S30202
> first. Use the OS79XX.TXT file to specify this version. After that
> upgrade to
> each newer release  P0S30203, P0S3-03-2-00, etc in the same fashion. Going
> from 6.3 to 7.0 the loader process changes. You need the OS79XX.TXT file,
> the SIP.cnf and SEP.cnf.xml for the phone. From 7.1 on you
> don't need the OS79XX.TXT file anymore.
> 
> 
> Nicolas Chabbey wrote:
> 
> >Hello,
> >
> >I've recently received a Cisco 7960G phone with the factory default
> >SCCP firmware on it.
> >As we're using SIP on our network, the first things i've done was to
> >upgrade but unfortunately the phone just restarted. By looking on the
> >TFTP logs and tcpump output, i've seen that the phone crashed and
> >restarted just after downloading the OS79XX.TXT file, without
> >requesting the image file at any moment.
> >
> >If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT
> >(begining with P003), the phone doesn't crash and request the
> >respective SEP.xml file. Unfortunately (again), just after
> >downloading the xml configuration it hang and restart. I've checked
> >the syntax and they's no error on it, if they's one the phone output
> >the error on the display without crashing. Note that i've both put
> >with and without the load information statement, with the same result.
> >
> >Both statical and DHCP configuration has been tried.
> >Maybe it's an hardware failure or i've miss somethings realy important :)
> >
> >Thanks
> >
> >-
> >Nicolas Chabbey <[EMAIL PROTECTED]>
> >Leafnet Networking Research Laboratory
> >http://www.bgp6.info
> >-
> >___
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> >
> 
> --
> 
> ISC Network Engineering
> The University of Pennsylvania
> 3401 Walnut Street, Suite 221A
> Philadelphia, PA 19104
> 
> voice: 215-573-8396
> 
>215-746-8001
> 
> fax: 215-898-9348
> 
> sip:[EMAIL PROTECTED]
> 
>
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Re: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade

[Nicolas Chabbey]

I've tried different versions, like the 5.0, 6.0 SIP but it doesn't
make any changes as the phone doesn't request the image, nor any
others file after downloading the AS79XX.TXT.
Once restarted, it do the same things, like configurating IP,
requesting the load file and configurations on the TFTP and looping
endless by restarting, crashing, restarting,...



On Thu, 3 Feb 2005 07:38:18 -0600, Matt Schulte <[EMAIL PROTECTED]> wrote:
> Which sip ver are you trying to install. Is it stuck in a loop or
> anything?
> 
> -----Original Message-
> From: Nicolas Chabbey [mailto:[EMAIL PROTECTED]
> Sent: Thursday, February 03, 2005 7:18 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade
> 
> Hello,
> 
> I've recently received a Cisco 7960G phone with the factory default SCCP
> firmware on it. As we're using SIP on our network, the first things i've
> done was to upgrade but unfortunately the phone just restarted. By
> looking on the TFTP logs and tcpump output, i've seen that the phone
> crashed and restarted just after downloading the OS79XX.TXT file,
> without requesting the image file at any moment.
> 
> If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT
> (begining with P003), the phone doesn't crash and request the respective
> SEP.xml file. Unfortunately (again), just after downloading the xml
> configuration it hang and restart. I've checked the syntax and they's no
> error on it, if they's one the phone output the error on the display
> without crashing. Note that i've both put with and without the load
> information statement, with the same result.
> 
> Both statical and DHCP configuration has been tried.
> Maybe it's an hardware failure or i've miss somethings realy important
> :)
> 
> Thanks
> 
> -
> Nicolas Chabbey <[EMAIL PROTECTED]>
> Leafnet Networking Research Laboratory
> http://www.bgp6.info
> -
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[Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade

[Nicolas Chabbey]

Hello,

I've recently received a Cisco 7960G phone with the factory default
SCCP firmware on it.
As we're using SIP on our network, the first things i've done was to
upgrade but unfortunately the phone just restarted. By looking on the
TFTP logs and tcpump output, i've seen that the phone crashed and
restarted just after downloading the OS79XX.TXT file, without
requesting the image file at any moment.

If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT
(begining with P003), the phone doesn't crash and request the
respective SEP.xml file. Unfortunately (again), just after
downloading the xml configuration it hang and restart. I've checked
the syntax and they's no error on it, if they's one the phone output
the error on the display without crashing. Note that i've both put
with and without the load information statement, with the same result.

Both statical and DHCP configuration has been tried.
Maybe it's an hardware failure or i've miss somethings realy important :)

Thanks

-
Nicolas Chabbey <[EMAIL PROTECTED]>
Leafnet Networking Research Laboratory
http://www.bgp6.info
-
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[Asterisk-Users] Compiling H323 channels with FC3 and RedhatSE3

[Nicolas FOURNIL]

Hello

I'm trying for a while to compile and install OH323 channels on my two
distribs...

I have downloaded the src pwlib and h323 files versions given in the
documentation.
Make some RPMS with "googled" SPECs (and seems to give good results)
Tried to compile the channels failed each time... (I have also tried
with at-rpms oh323 and pwlib versions).

Did someone who have already done the job could help me ? -I'm looking
for working specs to compile pwlib and oh323-

Thanks

Nicolas.


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[Asterisk-Users] DTMF skipped when calling from ISDN to SIP...

[Nicolas FOURNIL]

Hello

I have done the following test-network:

IP-Phone <=> ASTERISK <==> ISDN <> PSTN Phone
(SIP)  +
  SER


When I'm calling from the PSTN phone to the IP (SIP) phone:
I cannot get ANY DTMF from PSTN, they seem destroyed by the codec (small
scratches).
I listen DTMF from IP-Phone (SIP INBAND!)

When I'm calling from SIP phone to PSTN:
Same result, no PSTN => IP DTMF !

Any ideas ?

Thanks!


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RE: [Asterisk-Users] Mysql and Voicemail

[Nicolas FOURNIL]

Hello

try to enable mysql debug:

"log=/var/log/mysqlfull.log" in your /etc/my.cnf
and off course reload mysql

then

tail -f /var/log/mysqlfull.log

it will show you if your asterisk connects to the DB... if not, it's a
makefile problem... re-read tutorial...

PS: don't forget to try if your "full-log" works by connect by anyways to
the db. (And works fine like mysql ans "show databases" command)


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Alessio
Focardi
Envoye : mardi 28 decembre 2004 10:41
A : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Mysql and Voicemail


Hi,

I would like to enable mysql handling of voicemail boxes ... following
that tutorial

http://www.voip-info.org/wiki-Asterisk+voicemail+database

so I modified the makefile of /apps directory to include

USE_MYSQL_VM_INTERFACE=1

and copied mysql-vm-routines.h in the /apps dir, set up the db and
some boxes in the table, also edited the voicemail.conf file.

Everything compiles just fine, then when I start * I have no results,

"show voicemail users" --> There are no voicemail users currently defined

also if I try to check against a box with "MailboxExists" it does not
result created 

Any idea of what I'm getting wrong ?

tnx !




--
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] PA1688 Chipset IP Phones & ATA's

[Nicolas Moreau]

Hi

I updated on the phones here and at least it now logs on, use to only
work with SIP, but very quickly the host becomes unreachable.
I managed to place one call and the (Transf, HOLD & Flash keys) didn't
work either.

Nic



On Tue, 21 Dec 2004 08:55:06 +1000, Gary <[EMAIL PROTECTED]> wrote:
> On Mon, 20 Dec 2004 10:09:48 -0500, Kanuri, Seshu (Company IT) wrote:
> 
> >
> >
> >/Snip/
> >For those of you who may be interest
> >
> >IAX2 loads are now available for the standard builds...
> >
> >http://www.aredfox.com/edownloadsiax2.htm
> >
> >Just a word of caution...
> >
> >Remember to change the ports over to 4569 from whatever.
> >
> >And don't forget to grab the palmtool from
> >
> >http://www.aredfox.com/download/tools/PalmTool.zip
> >
> >My own testing of IAX2 with both a phone and an ATA is that IAX2 is
> >working very well :-)
> >
> >Gary
> >
> >/Snip/
> >
> >Gary,
> >
> >On the Centrality site there are  several  Zip files to download.
> >What are these different versions represent?
> >What does the Suffix F and B and T represent?
> >
> >PA168X Products IAX2 Program Download
> >
> >File Name Content Format Size Update
> >PA168F.zip PA168F VER 1.40 BIN 152K 2004/12/20
> >PA168FB.zip PA168FB VER 1.40 BIN 150K 2004/12/20
> >PA168FS.zip PA168FS VER 1.40 BIN 150K 2004/12/20
> >PA168T.zip PA168T VER 1.40 BIN 150K 2004/12/20
> >IP8008.zip IP8008 VER 1.40 BIN 150K 2004/12/20
> >PingHe.zip WuChuan VER1.40 BIN 150K 2004/12/20
> >5111PHONE.zip WuChuan VER1.40 BIN 386K 2004/12/20
> >PA168P.zip PA168P VER1.40 BIN 386K 2004/12/20
> >PA168Q.zip PA168Q VER1.40 BIN 386K 2004/12/20
> >PA168S.zip PA168S VER1.40 BIN 386K 2004/12/20
> >
> >Seshu Kanuri
> 
> The different versions are for the different versions of PA1688
> phones/ata's.
> 
> In my Case, the ip PHONE is identified by its label...
> 
> Type HOP-1002  Rev 004  << the rev is important)
> Hirakawa Electronics Co Ltd
> 
> Their home page is www.5111soft.com
> 
> As its a Rev 004 its actually a PingHE load.
> if it was a Rev 008 its a 5111Phone load.
> 
> I use their ATA unit as well its an AG-168
> 
> Looking at their home page (which in typical chinese fashion is in
> Chinglish, but note the download names...
> 
> The AG-168 uses a PA168Q load.
> 
> You must research your hardware !!
> 
> Just a tip for those playing with their phones ata/'s for the first
> time...
> 
> You have a good chance of screwingup your loading, particularly with
> a wrong version.. I personally recommend playing with phones over ata's
> at first recovery technique ...
> 
> remember what the factory default base addess is, in majority of cases
> its 192.168.1.100 but depending upon manufacturer & model it might
> be different.
> 
> Hold the * key down whilst turning on phone, You must do this twice
> (turn on's in succession). Presto, you are back at default.
> 
> Have fun,
> 
> Gary
> .
> 
> 
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[Asterisk-Users] VoiceMailMain login problem... new BUG ?

[Nicolas FOURNIL]

Hello

I think I've found a new bug, but first I'm asking for experts...

I have the following simple configuration :

in extensions.conf :
exten => 0660,1,VoicemailMain(${CALLERIDNUM})

So the caller is directly connected to his mailbox, it works great with
other users (like xlite, 0467161616, nfovdt...) but with the user pnunes :

When I tring to connect with the user "pnunes" I cannot enter into the
mailbox... it seems there is a mistake somewere  (see the folder who is
"nunespnunes" instead of "pnunes"). Any idea ?

*CLI> -- Executing VoiceMailMain("SIP/petitvillage-0813b1e0", "pnunes")
in new stack
-- Playing 'vm-login' (language 'en')
-- No username but # key pressed. Using CID 'pnunes'
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-helpexit' (language 'en')
-- Playing 'vm-options' (language 'en')
-- Recording the message
-- Playing 'vm-rec-busy' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing:  voicemail/default/nunespnunes/busy format: wav49,
0x80f0130
-- x=1, open writing:  voicemail/default/nunespnunes/busy format: gsm,
0x80ed3b8
-- x=2, open writing:  voicemail/default/nunespnunes/busy format: wav,
0x814ac70

PS: I'm using MYSQL Voicemail and the database seems correctly invoked :
 SELECT password,fullname,email,pager,options FROM users WHERE
context='default' AND mailbox='pnunes'

Thanks for advice

Nicolas
http://www.call.fr

PS: We're planning making a small page on our VideoVoicemail test, it works
perfectly at this moment...


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Re: [Asterisk-Users] TDMoE or IAX?

[Nicolas Bougues]

On Sun, Dec 19, 2004 at 06:56:19PM +1100, Eric Bishop wrote:
> Hi all,
> 
> Information on this topic seems a little scarce, so I thought I'd try
> the list
> 
> Apart from the the coolness factor can anyone explain to me in what
> situation one would use TDMoE rather than IAX for communication
> betwwen 2 Asterisk servers?
>

I thing that you're mostly better with IAX between 2 Asterisk
servers. TDMoE, however, is not limited to Asterisk. It's part of
zaptel.

You can use it to transport a TDM link over an Ethernet network (or
IP, with some kind of tunneling), and get it back as a TDM link on the
other side (with proper hardware).

-- 
Nicolas Bougues
Axialys Interactive
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[Asterisk-Users] Voicemail Video Attachement format

[Nicolas FOURNIL]

Hello

I have tested the Video-Voicemail feature with our SIP Hard phone, it works
great !

I'm trying to convert the h263 file (who cannot be played with an out of
stock Windows Media Player) to another format for email forwarding (mpeg or
another WMP recognised format)

Anyone has tried to ? (I have tried transcode or ffmpeg without success)

thanks for advice.

Nicolas

FOURNIL Nicolas
P2P manager
http://www.videotel.fr

PS: We actualy doing a french translation for voicemail prompts, with also
local changes (date format etc...) we will release it asap.


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[Asterisk-Users] Asterisk 1.0.3 and chan_capi ?

[Nicolas]

did asterisk 1.0.3 and chan_capi runs together ?
thx
nico


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[Asterisk-Users] Using meetme video mode with SIP ?

[Nicolas FOURNIL]

Hello

I'm planning using meetme module with SIP videophones (Videotel Hard
Phones), I have tried the meetme module with the v option but I haven't get
any video... can someone get the video mode running in meetme ?

Thanks a lot

Nicolas
SIP Hard VideoPhones
http://www.call.fr

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RE: [Asterisk-Users] SJPhone SIP Tab

[Nicolas FOURNIL]

Hello

I'm planning using meetme module with SIP videophones (Videotel Hard
Phones), I have tried the meetme module with the v option but I haven't get
any video... can someone get the video mode running in meetme ?

Thanks a lot

Nicolas


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Re: [Asterisk-Users] A new alternative to see who is online

[Nicolas]

Hi,
Sorry it is not planned.
Regards,
Nicolas
genuix wrote:
Hi,
googd work, 

could it be any chance to make it running on linux box ??
JFA
On Mon, 2004-11-22 at 09:16 +0100, Nicolas wrote:
Thanks for all your comment.
I will release the source code but not for the moment. I really need to 
clean the graphical part but right away I don't have enough time.

Regards,
Nicolas Bruxer
Lyle Giese wrote:
It does seem to work with ZAP channels and releasing the source would be a
great addition to Asterisk.  I can see some User Interface improvments that
could be made, but it  appears to be a great foundation to work from as the
basic functionality is there now.
Lyle
- Original Message - 
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Saturday, November 20, 2004 4:45 PM
Subject: Re: [Asterisk-Users] A new alternative to see who is online



Is there any chance that you might release the source code so that others
can improve upon your code?  I can see a real need for an application like
this.  I just wish it could be tweaked a little.

--
Jim Dossey Computer Services
-- Original message --
From: [EMAIL PROTECTED]

Hi all,
I have been facing about the problem to know who is online with asterisk
PBX.

However users wanted to see it right away, without launching any
application. As

I could not find any solution with IP phones and users were really
complaining,

I decided to write this little application that runs under windows and
stays on

screen.
It is not perfect, but it works and I think it can help other people.
Have a look:
http://mapage.noos.fr/~b.nico/
Regards,
Nicolas
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Re: [Asterisk-Users] A new alternative to see who is online

[Nicolas]

Hi,
	You are true for the moment, I only listen to messages and don't 
request anything to asterisk. So I only manage registration when a sip 
phone register or unregister while the leon is already launched.

Two possible solutions:
	-Havin a server that remember states which I want to avoid
	-asking asterisk when leon connects.(I didn't had time to explore this 
solution)

Jens Hansen wrote:
Great program, thanks!
only one question: when i reboot my pc i can't see who is online, until the
sip user re-register their clients at the server. leon only seems to update
his "online" status when a sip client connects.
Jens
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas
Sent: Monday, November 22, 2004 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A new alternative to see who is online
Thanks for all your comment.
I will release the source code but not for the moment. I really need to 
clean the graphical part but right away I don't have enough time.

Regards,
Nicolas Bruxer
Lyle Giese wrote:
It does seem to work with ZAP channels and releasing the source would be a
great addition to Asterisk.  I can see some User Interface improvments
that
could be made, but it  appears to be a great foundation to work from as
the
basic functionality is there now.
Lyle
- Original Message - 
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Saturday, November 20, 2004 4:45 PM
Subject: Re: [Asterisk-Users] A new alternative to see who is online



Is there any chance that you might release the source code so that others
can improve upon your code?  I can see a real need for an application like
this.  I just wish it could be tweaked a little.

--
Jim Dossey Computer Services
-- Original message --
From: [EMAIL PROTECTED]

Hi all,
I have been facing about the problem to know who is online with asterisk
PBX.

However users wanted to see it right away, without launching any
application. As

I could not find any solution with IP phones and users were really
complaining,

I decided to write this little application that runs under windows and
stays on

screen.
It is not perfect, but it works and I think it can help other people.
Have a look:
http://mapage.noos.fr/~b.nico/
Regards,
Nicolas
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Re: [Asterisk-Users] A new alternative to see who is online

[Nicolas]

Thanks for all your comment.
I will release the source code but not for the moment. I really need to 
clean the graphical part but right away I don't have enough time.

Regards,
Nicolas Bruxer
Lyle Giese wrote:
It does seem to work with ZAP channels and releasing the source would be a
great addition to Asterisk.  I can see some User Interface improvments that
could be made, but it  appears to be a great foundation to work from as the
basic functionality is there now.
Lyle
- Original Message - 
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Saturday, November 20, 2004 4:45 PM
Subject: Re: [Asterisk-Users] A new alternative to see who is online


Is there any chance that you might release the source code so that others
can improve upon your code?  I can see a real need for an application like
this.  I just wish it could be tweaked a little.
--
Jim Dossey Computer Services
-- Original message --
From: [EMAIL PROTECTED]
Hi all,
I have been facing about the problem to know who is online with asterisk
PBX.
However users wanted to see it right away, without launching any
application. As
I could not find any solution with IP phones and users were really
complaining,
I decided to write this little application that runs under windows and
stays on
screen.
It is not perfect, but it works and I think it can help other people.
Have a look:
http://mapage.noos.fr/~b.nico/
Regards,
Nicolas
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[Asterisk-Users] Re: can not compile chan_capi 0.3.5

[Nicolas]

Patrick wrote:

> On Sun, 2004-10-17 at 22:12, Nicolas wrote:
>> Hello,
>> 
>> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform.
>> i run latest asterisk cvs build 14/10/04.
> 
> Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk
> Works for me.
> 
> Regards,
> Patrick
> 
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Thanks but asterisk is working, my problem is the chan_capi channel driver.

nicolas


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[Asterisk-Users] can not compile chan_capi 0.3.5

[Nicolas]

Hello,

i can not compile chan_capi 0.3.5 on a suse 9.1 plattform.
i run latest asterisk cvs build 14/10/04.

just type make and become:

# make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g 
-I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686  -DCAPI_ES
-DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes
-Wno-missing-declarations -DCRYPTO   -c -o chan_capi.o chan_capi.c
In file included from /usr/include/linux/kernelcapi.h:13,
 from /usr/include/linux/capi.h:18,
 from chan_capi.c:35:
/usr/include/linux/list.h:604:2: warning: #warning "don't include kernel
headers in userspace"
chan_capi.c: In function `capi_new':
chan_capi.c:1073: error: structure has no member named `callerid'
chan_capi.c:1074: error: structure has no member named `dnid'
chan_capi.c: In function `pipe_msg':
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c:1724: error: structure has no member named `dnid'
chan_capi.c: In function `load_module':
chan_capi.c:2793: warning: passing arg 4 of `ast_channel_register' from
incompatible pointer type
make: *** [chan_capi.o] Error 1
#

A google search can not help.
can you help me ?

greetings
nicolas


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[Asterisk-Users] chan_capi install problem

[Nicolas Whitham]

Please can someone help me to install chan_capi on Mandrake 10.  I get page
after page of errors and can not seem to find detailed install instructions
anywhere.


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[Asterisk-Users] has anyone the capiCD() funktion in chan_capi running?

[Nicolas]

has anyone the capiCD() funktion in chan_capi running?
for me it does not do calldeflection.
the capi is messageing that CD is supported.
nico

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[Asterisk-Users] Do not get calldeflection (capiCD) to work.

[Nicolas]

I do not get calldeflection (capiCD) to work.

The Mobile do not ring, it seems the CD do not work.

I use the chan_capi 0.3.5 and have no idea.

please help me.
nico


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[Asterisk-Users] I have on isdn phone and one isdn-card (ast-pbx)attached (to my line)...

[Nicolas]

and i would happy if it be possible if the isdn-phone rings (number xyz) the
ast-pbx rings too (number xyz is configured in capi.conf too).

can anyone help me with that or is it not possible.
If the isdn-phone rings the ast-pbx do not.
Can it be that my phone answers the call without the call is taken?

nico


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[Asterisk-Users] Re: 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls

[Nicolas]

Hey where you buyed the isdn-cards? i tryed to get some without success.

nico

Marco Czudej wrote:

> Hello everyone,
> 
> I bought 2 HFC-ISDN Cards and want to run the first
> card in NT-Mode an the second one in TE-Mode.
> 
> Everything looks ok under SuSE 9.1, but I can't dial
> out.
> 
> I removed one card, for testing purposes and want to
> run this one card in TE-Mode. I only want to make a
> call with my Grandstream BT-101 over Asterisk via
> ISDN.
> 
> When I try to make a call I get:
> -
>  Executing Dial("SIP/11-3ef2", "Zap/g1/00MY-NUMBER")
> in new stack
> Aug 25 18:11:00 NOTICE[1117453232]: app_dial.c:727
> dial_exec: Unable to create channel of type 'Zap'
>   == Everyone is busy/congested at this time
> -
> 
> 
> "zap show channels" says:
> -
>   Chan Extension  Context Language
> MusicOnHold
>  pseudodefault
>   1default
>   2default
> -
> 
> 
> "ztcfg -vvv" tells:
> -
> Zaptel Configuration
> ==
> 
> SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
> 
> Channel map:
> 
> Channel 01: Individual Clear channel (Default)
> (Slaves: 01)
> Channel 02: Individual Clear channel (Default)
> (Slaves: 02)
> Channel 03: D-channel (Default) (Slaves: 03)
> 
> 3 channels configured.
> -
> 
> Only one point in "zttool" I don't really understand:
> -
> Current Alarms: No alarms.
>  Ã Ã
>Ã ÃSync Source:Internally clocked
>Ã  a  Ã
>Ã ÃIRQ Misses:   0
>Ã  a  Ã
>Ã ÃBipolar Viol: 0
>Ã  a  Ã
>Ã ÃTx/Rx Levels: 0/  0
>Ã  a  Ã
>Ã ÃTotal/Conf/Act:   3/  3/  0
> -
> Conf = configured or conflicted?
> 
> 
> I try a "Loop" but nothing happend.
> TxA, TxB etc. are empty, too.
> 
> 
> Can someone help me? - I really need some sample
> configs, too.
> 
> Which linux distribution runs smoothest with Asterisk?
> 
> 
> Thanks!
> Marco Czudej
> 
> 
> 
> 
> 
> 
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[Asterisk-Users] RE: SMP Performance

[Nicolas]

How do you setup your snoms?
i have 2 problems:

1. call waiting indicator do not running (busy tone on 2. call).
2. How can i monitor a line like a callmanager with the led on the
function-buttons.

Can you help ?

nicolas

mattf wrote:

> We are currently running 4 asterisk servers in production all running SMP
> and performance is better under SMP than non-SMP. Right now we are
> averaging just under 20,000 calls (both in and out) a day on those 4
> servers.
> 
> As for the "BEST" VOIP phone, that is certainly up for debate. Here are my
> opinions:
> Cisco phones work well but are expensive
> Polycom phones are extremely similar to the Ciscos but are much
> cheaper.
> 3com phones are tricky to set up with Asterisk
> Snom phones are very good but take some getting used to
> I don't know of many people who have successfully set up Nortel VOIP
> phones on asterisk
> Avaya as always is expensive for what you get
> Pingtel's are pretty but there are current and future support and
> compatibility issues so I've heard
> Mitel VOIP phones work but do not offer enough features to justify
> the cost right now
> Grandstream phones are cheap(enough said)
> Sipura Analog adapters are very configurable and much cheaper than
> Cisco ATA
> 
> Hope this helps.
> 
> MATT---
> 
> -Original Message-
> From: Tim Jackson [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 25, 2004 9:44 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] SMP Performance
> 
> 
> 25 should be the max ever. This machine used to be my testbed server. I
> may end up swapping it out later for a 1U IBM, but I just wanted to make
> sure that in the meantime it'd be able to handle what we are doing with
> it. We bought it "refurbished" for $600 about a year ago. I was just
> wondering about the SMP part, I've been told that it doesn't work well
> with SMP, and then I've been told it works fine. I just wanted a 2nd or
> 3rd opinion before I went ahead and implemented this. Another dumb
> question, I've gotten the idea that the best phones out there are the
> Cisco 7960s, any other good phones out there that are decently priced?
> Nortel? 3Com?
> 
> -Tim
> 
> -Original Message-
> From: mattf [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 25, 2004 8:43 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] SMP Performance
> 
> There is nothing wrong with running Asterisk on SMP. It runs quite well
> actually.
> 
> I'm assuming you just have the Quad Xeon 450mhz sitting around because
> you
> can't buy them new anymore, so it probably isn't costing you anything to
> use
> it. In which case it isn't a waste. If you are paying more than $800 for
> it,
> save it and just buy a new P4 for less. A $200 machine may not be able
> to
> handle 25 concurrent conversations, and may have some used or
> sub-standard
> parts in it, so that may not be the best choice.
> 
> You should be able to have upto 25 channels running on this machine no
> problem, How many maximum conversations do you forsee running
> concurrently
> at one time on this system?
> 
> MATT---
> 
> -Original Message-
> From: Matt Schulte [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 25, 2004 9:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] SMP Performance
> 
> 
> Meaning Asterisk won't/can't take advantage of the four CPU's? Or it's
> overkill for this scenario?
> 
> -Original Message-
> From: joachim [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 25, 2004 12:52 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SMP Performance
> 
> 
> 
> Send me the quad and i'll send you a 200$ pc to do this job.
> 
> The quad is heavily overpowered.
> 
> Joachim.
> 
> At 22:00 24/08/2004, you wrote:
>>content-class: urn:content-classes:message
>>Content-Type: multipart/alternative;
>> boundary="_=_NextPart_001_01C48A15.130BF232"
>>
>>We're looking at implementing Asterisk in our department in the near
>>future, we're looking at anywhere from 15-25 extensions. The machine we
> 
>>were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache)
> w/
>>1GB of ram. I've heard bad things about running Asterisk on SMP
> machines?
>>Would we be running into any performance issues with this machine?
>>
>>Tim Jackson
>>Network Engineer
>>Angelina County, Te

[Asterisk-Users] I have on isdn phone and one isdn-card (ast-pbx) attached (to my line)...

[Nicolas]

and i would happy if it be possible if the isdn-phone rings (number xyz) the
ast-pbx rings too (number xyz is configured in capi.conf too).

can anyone help me with that or is it not possible.
If the isdn-phone rings the ast-pbx do not.
Can it be that my phone answers the call without the call is taken?

nico


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Re: [Asterisk-Users] Distinctive Ring Cadences

[Nicolas Gudino]

On Wed, 2004-08-25 at 20:38, Chris Shaw wrote:
> Cool! I could see this being very useful, for example you could have an IVR
> that says something like "Please set the priority of your call, 1 for
> urgent, 2 for normal or 3 for low" then if 1, bellcore-r4, if 2 bellcore-r3,
> if 1 bellcore-r1!

What for? People will allways hit 1 

-- 
Nicolas Gudino <[EMAIL PROTECTED]>
House Internet S.R.L.

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Re: [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones

[Nicolas Gudino]

t; > To UNSUBSCRIBE or update options visit:
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> >
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-- 
Nicolas Gudino <[EMAIL PROTECTED]>
House Internet S.R.L.

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Re: [Asterisk-Users] Error compiling meetme2

[Nicolas Gudino]

Hi Geoff,
Geoff Nordli wrote:

I was able to compile the module and it loads correctly, but I am still
having problems with the app.
I see all the users in the conference, but I can't kick them out, or change
their mode from talk to listen-and-talk.  No errors are showing up anywhere.
I am not really sure how to troubleshoot this, any ideas?
Thanks,
Enable register_globals in php.
You can also put an "extract($_GET);" in the top of the php file.
--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Multiple SIP phones ringing for same extension

[Nicolas Gudino]

Hi,
David Gurr wrote:
Can someone confirm what I should expect the correct behaviour to be on
incoming calls if I have multiple SIP phones configured for the same
username?
I'd expect all the phones registered under the username that that extension
is associated with to ring, and the first one that answers gets it.
What I get, is just the first phone that registered gets a ring. The second
one doesn't ring at all.
Asterisk won't work this way. Just the last phone registered will ring. 
There was a big thread a month ago in this list and a Bounty placed for 
adding the feature. Search for "sip simultaneous" in google or the wiki 
(http://www.voip-info.org)

--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Call stealing

[Nicolas Gudino]

Ben Merrills wrote:
Hi,
 

How can I (through the manager interface) steal a call from one phone, 
and transfer it to another? Does asterisk provide for actions like this? 
It’s a common action in Lucent systems it seems.

 

Cheers,
 

Ben
You can use the "Redirect" command. Visit http://www.asternic.org and 
look at the Flash Operator Panel. It can do that and more..

--
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House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Help - is voip good for in-house calls?

[Nicolas Gudino]

Andrew Kohlsmith wrote:
On Sunday 15 August 2004 12:03, Nicolas Gudino wrote:
If you already have the analog telephone wiring in place, and you are on
a budget, I recomend you to use sipura spa-2000 adapters. They are a
whole lot better than GS phones. You can have 3way conferences and
attendant transfers. With GS you cannot do that. The price is as good
for the sipuras as the GS phones, about $50 per FXS port, plus a cheap
analog phone and you will be all set.

Why on earth would you install SPA-2000s and endure that wiring mess?  An FXS 
channel bank and a BIX strip will save you YEARS in lost time due to wiring 
and general messiness!

I prefer the wiring mess and sipuras than the GS phones. That's all.
--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Help - is voip good for in-house calls?

[Nicolas Gudino]

Hi Francis,
Francis Augusto Medeiros wrote:
Hi there everyone!
I work at an office where we plant to have about 12-15 phone
extensions. Costs of PBX are cheaper, but they are not expandable and,
as the office is brand new, I want to use all modern stuff.
My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and
install and asterisk server, as well as a Digium TDM400 for POTS
access, will I have the same voice quality and standards as a
PBX-only, with "traditional" phones? Or should I go all the way to
Digium's TDM? Or should I forget the whole thing and get a traditional
PBX? ;)
If you already have the analog telephone wiring in place, and you are on 
a budget, I recomend you to use sipura spa-2000 adapters. They are a 
whole lot better than GS phones. You can have 3way conferences and 
attendant transfers. With GS you cannot do that. The price is as good 
for the sipuras as the GS phones, about $50 per FXS port, plus a cheap 
analog phone and you will be all set.

My concerns are most latencies. Our network will be a switch with lots
of ports, all 100mb/s, with VERY low traffic.
Internal calls (SIP to SIP) will sound great. You will probably 
experience some echo when going to POTS. I did not try the Sipura 
SPA-3000 yet, but it seems to be a cheap alternative to a gateway, 
providing you with one FXO and one FXS for $130 or so. the echo 
cancellation in the sipura works well for fxs, it might work well to for 
fxo.

--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] E1 monochannel :-(

[Nicolas Gudino]

Hola Horacio,
Comentarios en línea...
Horacio J. Peña wrote:
Hola!
I'm using asterisk as H.323 -> PRI gateway. First call goes
thru ok, second concurrent call fails with:
Aug  6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending  to 
context [ip2pri]
-- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new 
stack
Aug  6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of 
type 'Zap'
Aug  6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of 
type 'Zap'
== Everyone is busy at this time
Aug  6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in 
context 'ip2pri'
Aug  6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in 
context 'ip2pri'
Aug  6 11:52:40 DEBUG[81926]: chan_h323.c:1179 cleanup_connection: Cleaning up our mess
My configs are:
h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
gatekeeper = DISABLE
context=ip2pri
[ip2pri] ; is this needed?
type=user
context=ip2pri
extensions.conf:
[general]
static=yes
writeprotect=yes
[globals]
[ip2pri]
exten => _9.,1,Dial(Zap/1/${EXTEN:0}) ; i must send the 9 to the PRI...
^^^
Replace Zap/1 with Zap/g1
Saludos!
--
Nicolás Gudiño
House Internet S.R.L.
Buenos Aires - Argentina
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