date:20050905

In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
 
 Tony Mountifield wrote:
 
 In article [EMAIL PROTECTED],
 Doug Lytle [EMAIL PROTECTED] wrote:
   
 
 Anybody having issues with ztdummy under the current 2.6 RC7?  I get the 
 following errors when trying to modprobe ztdummy:
 
 Unable to register zaptel rtc driver
 
 Doing a Google on the error shows reference to a message from 2004 that 
 said you might not have RTC compiled into the kernel.  Checking via:
 
 cd /usr/src/linux-2.6.13-rc7
 grep -i rtc .config
 
 shows:
 
 CONFIG_APM_RTC_IS_GMT=y
 CONFIG_RTC=m
 CONFIG_GEN_RTC=m
 CONFIG_GEN_RTC_X=y
 CONFIG_HPET_RTC_IRQ=y
 CONFIG_SENSORS_RTC8564=m
 CONFIG_SND_RTCTIMER=m
 
 
 Any suggestions?
 
 rtc and genrtc are alternatives to each other.
 
 Make sure that the rtc module is loaded, and *not* genrtc.
 
 ztdummy is not compatible with genrtc, only with rtc.
 
 I had time tonight to try this.  Under Linux 2.6.13 final.  Looking at 
 make menuconfig shows that both Generic /dev/rtc emulation and Enhanced 
 Real Time Clock support
 
 Removing one and enabling the other, compiling and recompiling zaptel:
 
 make clean;make linux26 make install (udev rules in place)
 
 I am unable to do a modprobe ztdummy without the above error.  Any 
 others running Linux 2.6.13 and successfully using ztdummy for timing?

There was nothing wrong with the original kernel config, as both rtc and
genrtc were set to be compiled as modules.

What you need to do is find where the system is deciding to load genrtc
and make it load rtc instead. Failing that, before loading zaptel and
ztdummy, do modprobe -r genrtc followed by modprobe rtc.

Cheers
Tony
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Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] DTMF issue on IVR

2005-09-05 Thread larry lin


Hi All,

 I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with 
RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt 
and asks caller to dial four-digit extension. Caller has to dial slowly, 
otherwise, Asterisk cannot recognize the extension number. I look at the 
trace on Asterisk CLI and there are missing digit in the middle of string. 
ex, caller dials 3114, I can see 314 or 34 on CLI. I think the Asterisk 
barge-in response is vary slow, it usually takes half a second or so for the 
voice prompt to stop after the first key is hit. If a second key is hit when 
the prompt is still playing, this key will be missed and will not feed into 
the Asterisk IVR. However, Asterisk will be able to recognize all the keys 
if you wait long enough between the 1st key and 2nd key (and you can hit as 
fast as you can between 2nd, 3rd  4th key).
  I searched the wiki and did not find any related information. Is there a 
way to set the barge-in response time or how many keys you can buffer before 
the prompt is stopped ?


  Please advise,

 Thanks and Regards,

Larry


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[Asterisk-Users] No DID on ZAP

2005-09-05 Thread Darren Wright

I can't seem to get any ZAP trunks on my TE110P to match any extensions
for incoming DID.


I've even used the exten = _X.,1And it still will not match that.
All I get is:


 -- Starting simple switch on 'Zap/1-1'
  == Starting Zap/1-1 at zap-custom,s,1 failed so falling back to exten
's'
  == Starting Zap/1-1 at zap-custom,s,1 still failed so falling back to
context 'default'

The only think it will match is exten = s,1

And then it works fine...all Callerid is perfect.

Any ideas?

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Re: [Asterisk-Users] DTMF issue on IVR

2005-09-05 Thread maka

hya,

try using relaxdtmf=yes in zapata.conf and see if that solves it.
checkout these recent postings as well:

http://lists.digium.com/pipermail/asterisk-users/2005-August/122737.html
http://lists.digium.com/pipermail/asterisk-users/2005-August/122656.html

cheersOn 9/5/05, larry lin [EMAIL PROTECTED] wrote:
Hi All,I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 withRedHat 9. When the caller calls into our IVR, and IVR plays the first promptand asks caller to dial four-digit extension. Caller has to dial slowly,
otherwise, Asterisk cannot recognize the extension number. I look at thetrace on Asterisk CLI and there are missing digit in the middle of string.ex, caller dials 3114, I can see 314 or 34 on CLI. I think the Asterisk
barge-in response is vary slow, it usually takes half a second or so for thevoice prompt to stop after the first key is hit. If a second key is hit whenthe prompt is still playing, this key will be missed and will not feed into
the Asterisk IVR. However, Asterisk will be able to recognize all the keysif you wait long enough between the 1st key and 2nd key (and you can hit asfast as you can between 2nd, 3rd 4th key). I searched the wiki and did not find any related information. Is there a
way to set the barge-in response time or how many keys you can buffer beforethe prompt is stopped ? Please advise,Thanks and Regards,Larry___
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RE: [Asterisk-Users] hints and polycom IP 300 phones

2005-09-05 Thread harry gaillac

Hello,

I have two polycom ip300.
I patched Asterisk However it don't show status of
phones when I press busy, Away, ...

So I use Sip Express Router (proxy sip) for IM and
Presence SIMPLE.

Harry

--- Adam Goryachev
[EMAIL PROTECTED] a écrit :

 Hi all,
 
 I've just updated to current CVS, and have 2 polycom
 IP phones, one is a
 IP600 and the other is a IP300. The IP600 shows the
 status of the IP300
 and a ZAP line quite nicely, but the IP300 won't
 show the status of the
 IP600
 
 Is there any additional debug apart from show
 hints to see why this
 might not be working ??
 -= Registered Asterisk Dial Plan Hints =-
655 : SIP/gs102_1   State
  0 Watchers  0
605 : Zap/127   State
  0 Watchers  3
604 : SIP/ata186_2  State
  0 Watchers  0
603 : SIP/ata186_1  State
  0 Watchers  0
602 : Zap/129   State
  0 Watchers  0
601 : SIP/polycom_b State
  0 Watchers  1
600 : SIP/polycom_a State
  1 Watchers  2
 
 The IP600 is watching 605 and 600 and working nicely
 for both, the IP300
 is watching 601, but isn't working
 
 Has anyone got a IP300 phone to display the status
 ?? Any suggestions
 for things to look at/etc ??
 
 PS, of course, the current state is that 600 is
 off-hook and all others
 are on-hook.
 
 Regards,
 Adam
 
 -- 
  -- 
 Adam Goryachev
 Website Managers
 Ph:  +61 2 8304    
 [EMAIL PROTECTED]
 Fax: +61 2 8304 0001   
 www.websitemanagers.com.au
 
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Tr: [Asterisk-Users] MWI - message waiting indication

2005-09-05 Thread harry gaillac

Remarque : message transféré en pièce jointe.







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hello,

I read
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large


anybody could tell me more about this ?
Is it available with ARA ?

Regards
Harry


Method 3

Q: If you have your SIP phones registered with SER but
your voicemail is handled by asterisk, how do you get
the MWI (Message Waiting Indicator) light to function
on the phone?

A: In sip.conf create a section pointing at your SER
router.

 [ser]
 type=friend; We allow incoming and
outgoing calls. Use peer if you are only doing MWI
 context=ser; This is the context
incoming calls land in
 host=ser.server.tld; This is the hostname or
IP address of your SER server
 fromdomain=ser.server.rld  ; This is your SER_DOMAIN
 insecure=very  ; This allows incoming
calls from the phones routing through ser to be passed
into asterisk
 [EMAIL PROTECTED]   ; This is where you list
the voicemail boxes to monitor

This tells asterisk that if a voicemail comes in to
user then it needs to send a SIP NOTIFY message to
the ser.server.tld phone. Well this is all well and
good except how does SER deliver this NOTIFY to the
phones? First thing is that you need to make a tiny
change to the asterisk code to pass the mailbox user
in the SIP NOTIFY packet.

--- channels/chan_sip.c.origThu Jul 14 12:03:18
2005
+++ channels/chan_sip.c Thu Jul 14 12:05:26 2005
@@ -9710,6 +9710,7 @@
/* Called with peerl lock, but releases it */
struct sip_pvt *p;
int newmsgs, oldmsgs;
+   char *s;

/* Check for messages */
ast_app_messagecount(peer-mailbox, newmsgs,
oldmsgs);
@@ -9735,6 +9736,10 @@
/* Recalculate our side, and recalculate Call
ID */
if
(ast_sip_ouraddrfor(p-sa.sin_addr,p-ourip))
memcpy(p-ourip, __ourip,
sizeof(p-ourip));
+   strcpy(p - username, peer - mailbox);  /*
Username = Mailbox name */
+   s = strchr(p - username, '@');  /*
Remove the context part */
+   if (s != NULL)
+*s = 0;
build_via(p, p-via, sizeof(p-via));
build_callid(p-callid, sizeof(p-callid),
p-ourip, p-fromdomain);
/* Send MWI */



After this patch is applied, the MWI NOTIFY messages
coming from asterisk will have the URI
[EMAIL PROTECTED] This can be then routed with ser
to the correct phone with normal SER routing rules.
ie. SER does a lookup(location) and then a
t_relay(). I don't believe this patch should effect
any non-ser controlled sip phones.

For me, this method was a lot easier then Method 2
listed above. You can add as may mailbox's as you like
into the mailbox= line in the asterisk sip.conf file.
One possible problem is if you have a mailbox called
[EMAIL PROTECTED] and another called [EMAIL PROTECTED], this patch will
make the MWI indicator light up for phone
[EMAIL PROTECTED] when either mailbox gets a
message. A simple modification to the patch and SER
could be used to handle multiple contexts if required
however this simplification is sufficient for me.








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[Asterisk-Users] GotoIf sample...

2005-09-05 Thread ryan nalupa

hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errorsthanks! : )__Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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[Asterisk-Users] queue transfers always get EXITWITHKEY

2005-09-05 Thread lenz



Hello list,
while using the new Asterisk 1.2 beta, I keep noticing this when an agent  
transfers a call from a queue to another extension:


[Except from queue_log]
1125912636|1125912630.134|queue-dps|NONE|ENTERQUEUE||21
1125912638|1125912630.134|queue-dps|Agent/101|CONNECT|2
1125912641|1125912630.134|queue-dps|Agent/101|TRANSFER|22|sip
1125912641|1125912630.134|queue-dps|NONE|EXITWITHKEY||1

As soon as the call is transfered, the queue system logs an EXITWITHKEY  
that I do not understand. It has always the same timestamp as the TRANSFER  
line. The call keeps going correctly as I would expect and is not hang up,  
while the agent gets back to being free. Anybody knows what the extra  
EXITWITHKEY means?

Thanks
l.



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Re: [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working

Here you go, eagerly awaiting comments:

-- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack
-- Executing Dial(SIP/xlite1-e0a7, CAPI/hfcpci/17/b) in new stack
data = hfcpci/17/b
parsed dialstring: 'hfcpci' '17' 'b'
capi request for interface 'hfcpci'
parsed dialstring: 'hfcpci' '17' 'b'
  == hfcpci: Call CAPI/hfcpci/17-0 with B3  (pres=0x00, ton=0x00)
CONNECT_CONF ID=001 #0x0003 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

-- hfcpci: received CONNECT_CONF PLCI = 0x101
CONNECT_REQ ID=001 #0x0003 LEN=0044
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x1
  CalledPartyNumber   = 8017
  CallingPartyNumber  = 00 800
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= 00 00
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- Called hfcpci/17/b
INFO_IND ID=001 #0x0001 LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x8
  InfoElement = 81 81

INFO_RESP ID=001 #0x0001 LEN=0012
  Controller/PLCI/NCCI= 0x101
-- hfcpci: info element CAUSE 81 81
DISCONNECT_IND ID=001 #0x0002 LEN=0014
  Controller/PLCI/NCCI= 0x101
DISCONNECT_IND ID=001 #0x0002 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Reason  = 0x3481

DISCONNECT_RESP ID=001 #0x0002 LEN=0012
  Controller/PLCI/NCCI= 0x101

CAPI INFO 0x3481: Unallocated (unassigned) number
  == hfcpci: CAPI Hangingup
  == hfcpci: Interface cleanup PLCI=0x101
  == No one is available to answer at this time (1:0/0/0)
  == Auto fallthrough, channel 'SIP/xlite1-e0a7' status is 'NOANSWER'



On 04/09/05, Armin Schindler [EMAIL PROTECTED] wrote:
 This is not enough to see the problem.
 Use verbose level 5 (-v)
 and use 'capi debug'
 
 Armin
 
 On Sun, 4 Sep 2005, Konrads Smelkovs wrote:
  See if this helps... , i ran asterisk with -vvvgc
 
  CAPI Debugging Enabled
  -- Executing SetCallerID(SIP/xlite1-1be2, 0) in new stack
  -- Executing Dial(SIP/xlite1-1be2, CAPI/hfcpci/17/b) in new stack
  data = hfcpci/17/b
  capi request for interface 'hfcpci'
== hfcpci: Call CAPI/hfcpci/17-1 with B3  (pres=0x00, ton=0x00)
  -- hfcpci: received CONNECT_CONF PLCI = 0x101
  -- Called hfcpci/17/b
== hfcpci: Interface cleanup PLCI=0x101
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'SIP/xlite1-1be2' status is 'NOANSWER'
 
  Maybe there is something more I could look for?
  On 02/09/05, Armin Schindler [EMAIL PROTECTED] wrote:
   On Fri, 2 Sep 2005, Konrads Smelkovs wrote:
Hello,
These are error messages I get when I try to call a number over CAPI 
channel.
   
-- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack
-- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack
data = hfcpci/b17
capi request for interface 'hfcpci'
  == hfcpci: Call CAPI/hfcpci/b17-1   (pres=0x00, ton=0x00)
-- hfcpci: received CONNECT_CONF PLCI = 0x201
-- Called hfcpci/b17
  == hfcpci: Interface cleanup PLCI=0x201
  == No one is available to answer at this time (1:0/0/0)
  == Auto fallthrough, channel 'SIP/xlite1-3b80' status is 'NOANSWER'
   
   
mISDNUser test tools show ISDN line working (testcon).
   
capi info shows that 2 B channels are available
capiinfo utility also dumps meaningful information - indicating that
it indeed recognises the card.
  
   To see more, you may want to increase verbosity level and enable
   'capi debug'.
  
   Anyway, if you are using CVS version of chan_capi, your dialstring is not
   correct. The option for earlyb3 'b' may not be part of the called number 
   any
   more. Option are added after the called id and an additional '/'.
   Your dial command should look like this:
Dial(CAPI/hfcpci/17/b)
  
chan_capi version - CVS as of Sep 2, 2005 from chan-capi.sf.net
I read comments on voip-info about 2.6.12 kernel breaking something,
but the patch was for capi 0.3.5, not sure it applies...
  
   No, this is obsolete for chan_capi on sourceforge. No patches are needed.
  
   Armin
  
  
 
 
  --
  Konrads Smelkovs
  Applied IT sorcery.
 
 


-- 
Konrads Smelkovs
Applied IT sorcery.
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SV: [Asterisk-Users] sending fax

2005-09-05 Thread Arne Morten Johansen

What about faxing yourself if you don't have a scanner? 

-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Johan van Tongeren
Sendt: 5. september 2005 09:11
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: RE: [Asterisk-Users] sending fax

[macro-fax-dialing]
exten = s,1,SetCIDNum(0${CALLERIDNUM})
exten = s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t)
exten = s,3,Goto(900)
exten = s,103,Goto(900)
exten = s,900,Busy
exten = s,901,Hangup

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Chris Shipman
Verzonden: maandag 5 september 2005 7:22
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] sending fax

I've read alot on the wiki about sending and receiving faxes thru
asterisk.
I've gotten the receive to work great.My question is how does one
send a
fax?
I see lots of instructions about how to send the image to asterisk by
email,
etc.  The problem is how does  one make the image of the fax to
begin
with?   Has anyone come up with a good solution for this?


Regards,


Chris


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Re: [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7

2005-09-05 Thread Doug Lytle



Tony Mountifield wrote:


In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
 


There was nothing wrong with the original kernel config, as both rtc and
genrtc were set to be compiled as modules.

What you need to do is find where the system is deciding to load genrtc
and make it load rtc instead. Failing that, before loading zaptel and
ztdummy, do modprobe -r genrtc followed by modprobe rtc.

 



Thanks for the input Tony, but the instructions that Rob Thomas wrote 
took care of my issue.


Thanks again to both of you!

Doug


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RE: [Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7

2005-09-05 Thread Rob Thomas

 -Original Message-
 Thanks for the input Tony, but the instructions that Rob Thomas wrote
 took care of my issue.
 
 Thanks again to both of you!

You're welcome, Happy to help.

--Rob

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Re: [Asterisk-Users] kernel panic

2005-09-05 Thread Tzafrir Cohen

On Sun, Sep 04, 2005 at 07:10:29PM -0600, Michael Welter wrote:
 I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates). 
  The system has a TE110P card, and zaptel.conf is configured for an E1.
 
 
 When I do a 'zaptel stop' I get a kernel panic.

Did you stop asterisk first?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] Nokia 32 Terminal

2005-09-05 Thread Sergio Chersovani


AbdelRahman Tarzi ha scritto:


If you wish to connect it to an FXS you will need a special cable which
Nokia sells..
 

you don't really need a special cable for FXS, the cable is a standard 
phone cable with a j11 4/6 pin plug. Just read the tech manual from the 
nokia website for the pinout.



Connecting to an FXO (which expects a line) is the default.
Check the normal stuff (like dialstring) before you suspect the device..
They're really maintenance-free !!
 

I have a problem with the external antenna. No signal gain with it 
connected to the nokia 32 terminal.
You can play with the AT command via serial port to see the signal 
quality level, and I advice you to disable the gsm call waiting service.


Sergio
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Re: [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working

Hello, 
I solved the problem - i was setting wrong caller-ID and thus got rejected.
Thanks for help.

On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote:
 Here you go, eagerly awaiting comments:
 
 -- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack
 -- Executing Dial(SIP/xlite1-e0a7, CAPI/hfcpci/17/b) in new stack
 data = hfcpci/17/b
 parsed dialstring: 'hfcpci' '17' 'b'
 capi request for interface 'hfcpci'
 parsed dialstring: 'hfcpci' '17' 'b'
   == hfcpci: Call CAPI/hfcpci/17-0 with B3  (pres=0x00, ton=0x00)
 CONNECT_CONF ID=001 #0x0003 LEN=0014
   Controller/PLCI/NCCI= 0x101
   Info= 0x0
 
 -- hfcpci: received CONNECT_CONF PLCI = 0x101
 CONNECT_REQ ID=001 #0x0003 LEN=0044
   Controller/PLCI/NCCI= 0x1
   CIPValue= 0x1
   CalledPartyNumber   = 8017
   CallingPartyNumber  = 00 800
   CalledPartySubaddress   = default
   CallingPartySubaddress  = default
   BProtocol
B1protocol = 0x1
B2protocol = 0x1
B3protocol = 0x0
B1configuration= default
B2configuration= default
B3configuration= default
   BC  = default
   LLC = default
   HLC = default
   AdditionalInfo
BChannelinformation= 00 00
Keypadfacility = default
Useruserdata   = default
Facilitydataarray  = default
 
 -- Called hfcpci/17/b
 INFO_IND ID=001 #0x0001 LEN=0017
   Controller/PLCI/NCCI= 0x101
   InfoNumber  = 0x8
   InfoElement = 81 81
 
 INFO_RESP ID=001 #0x0001 LEN=0012
   Controller/PLCI/NCCI= 0x101
 -- hfcpci: info element CAUSE 81 81
 DISCONNECT_IND ID=001 #0x0002 LEN=0014
   Controller/PLCI/NCCI= 0x101
 DISCONNECT_IND ID=001 #0x0002 LEN=0014
   Controller/PLCI/NCCI= 0x101
   Reason  = 0x3481
 
 DISCONNECT_RESP ID=001 #0x0002 LEN=0012
   Controller/PLCI/NCCI= 0x101
 
 CAPI INFO 0x3481: Unallocated (unassigned) number
   == hfcpci: CAPI Hangingup
   == hfcpci: Interface cleanup PLCI=0x101
   == No one is available to answer at this time (1:0/0/0)
   == Auto fallthrough, channel 'SIP/xlite1-e0a7' status is 'NOANSWER'
 
 
 
 On 04/09/05, Armin Schindler [EMAIL PROTECTED] wrote:
  This is not enough to see the problem.
  Use verbose level 5 (-v)
  and use 'capi debug'
 
  Armin
 
  On Sun, 4 Sep 2005, Konrads Smelkovs wrote:
   See if this helps... , i ran asterisk with -vvvgc
  
   CAPI Debugging Enabled
   -- Executing SetCallerID(SIP/xlite1-1be2, 0) in new stack
   -- Executing Dial(SIP/xlite1-1be2, CAPI/hfcpci/17/b) in new stack
   data = hfcpci/17/b
   capi request for interface 'hfcpci'
 == hfcpci: Call CAPI/hfcpci/17-1 with B3  (pres=0x00, ton=0x00)
   -- hfcpci: received CONNECT_CONF PLCI = 0x101
   -- Called hfcpci/17/b
 == hfcpci: Interface cleanup PLCI=0x101
 == No one is available to answer at this time (1:0/0/0)
 == Auto fallthrough, channel 'SIP/xlite1-1be2' status is 'NOANSWER'
  
   Maybe there is something more I could look for?
   On 02/09/05, Armin Schindler [EMAIL PROTECTED] wrote:
On Fri, 2 Sep 2005, Konrads Smelkovs wrote:
 Hello,
 These are error messages I get when I try to call a number over CAPI 
 channel.

 -- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack
 -- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new 
 stack
 data = hfcpci/b17
 capi request for interface 'hfcpci'
   == hfcpci: Call CAPI/hfcpci/b17-1   (pres=0x00, ton=0x00)
 -- hfcpci: received CONNECT_CONF PLCI = 0x201
 -- Called hfcpci/b17
   == hfcpci: Interface cleanup PLCI=0x201
   == No one is available to answer at this time (1:0/0/0)
   == Auto fallthrough, channel 'SIP/xlite1-3b80' status is 'NOANSWER'


 mISDNUser test tools show ISDN line working (testcon).

 capi info shows that 2 B channels are available
 capiinfo utility also dumps meaningful information - indicating that
 it indeed recognises the card.
   
To see more, you may want to increase verbosity level and enable
'capi debug'.
   
Anyway, if you are using CVS version of chan_capi, your dialstring is 
not
correct. The option for earlyb3 'b' may not be part of the called 
number any
more. Option are added after the called id and an additional '/'.
Your dial command should look like this:
 Dial(CAPI/hfcpci/17/b)
   
 chan_capi version - CVS as of Sep 2, 2005 from chan-capi.sf.net
 I read comments on voip-info about 2.6.12 kernel breaking something,
 but the patch was for

[Asterisk-Users] [EMAIL PROTECTED] and zaphfc dial out not working

2005-09-05 Thread [EMAIL PROTECTED]

Hello,

 I have [EMAIL PROTECTED] with zaphfc patch applied
 (http://dondisperato.blogspot.com/), but I can not make call to 
 legacy PBX (Alcatel 4400). I can only accept incoming calls.

 I am dialing with this: exten = 202,1,Dial(Zap/g1/242)

---
 asterisk1*CLI bri debug span 1
Enabled debugging on span 1
-- Executing Dial(SIP/201-4678, Zap/g1/242) in new stack
-- Making new call for cr 131
-- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=26
 Call Ref: len= 1 (reference 3/0x3) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
(16)
  Ext: 1  User information layer 1: A-Law (35)
 [18 01 81]
 Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0, Preferred 
Dchan: 0
ChanSel: B1 channel
 ]
 [6c 05 00 80 32 30 31]
 Calling Number (len= 7) [ Ext: 0  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0)
   Presentation: Presentation permitted, user number 
not screened (0) '201' ]
 [70 04 80 32 34 32]
 Called Number (len= 6) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0) '242' ]
 [a1]
 Sending Complete (len= 1)
-- Called g1/242
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 1 (reference 131/0x83) (Terminator)
 Message type: RELEASE COMPLETE (90)
 [08 03 81 95 80]
 Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: 
Private network serving the local user (1)
  Ext: 1  Cause: Call Rejected (21), class = Normal Event 
(1) ]
  Cause data 1: 80 (128)
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
-- Zap/1-1 is circuit-busy
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time
--
--
zaptel.conf:
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=nl
defaultzone=nl

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

--
zapata.conf:
;
; Zapata telephony interface
;
; Configuration file
;
[channels]
musiconhold  = default
language = en
;
; ISDN
;
switchtype= euroisdn
echocancel= yes
immediate = no
overlapdial   = no
pridialplan   = unknown
prilocaldialplan  = unknown
nationalprefix= 0
internationalprefix = 00

context   = from-pstn
signalling= bri_cpe_ptmp  ; HFC-S TE mode
usecallerid   = yes
usecallingpres= yes
group = 1
channel = 1-2

 what is wrong?
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[Asterisk-Users] Asterisk clustering with SIP proxy?

2005-09-05 Thread Roy Sigurd Karlsbakk


hi

i've heard it should be possible, but i can't find out how...

I want to configure a bunch of asterisk boxes to do SIP/PSTN  
connectivity, and I need SER or something to do some balancing in  
front of them. The requirements are listed below.


 * SER MUST accept and load balance incoming calls over n asterisk  
boxes (anything between 2 and 20 servers depending on installation)
 * If SER forwards a call to a server being busy or down, SER SHOULD  
retry on another server
 * SER SHOULD balance the number of calls to each server based on  
the codec used so single servers will not be overloaded by  
transcoding costs.
 * If possible, SER SHOULD be able to fail over to another SER box  
if SER fails.


Does anyone know if this is possible? I'd gladly pay someone to help  
me out here...


roy
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Re: [Asterisk-Users] kernel panic

2005-09-05 Thread Elio Rojano





Sure yourself that your card haven't IRQ shared.
In this case (you have IRQ conflict) change your card of PCI slot, or
modify IRQ assignment on BIOS and try again unload wcte11xp/zaptel
drivers.

Tzafrir Cohen wrote:

  On Sun, Sep 04, 2005 at 07:10:29PM -0600, Michael Welter wrote:
  
  
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates). 
 The system has a TE110P card, and zaptel.conf is configured for an E1.


When I do a 'zaptel stop' I get a kernel panic.

  
  
Did you stop asterisk first?

  




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[Asterisk-Users] asterisk CAPI dial-in issues

Hello configuration as follows, dial-out works:

capi.conf:
[hfcpci]
;;PointToPoint (55512-0)
isdnmode=MSN
incomingmsn=*
;msn=61
controller=1
devices=2
context=incoming

extensions.conf:
[incoming]
exten = _XX,1,Playback(demo-abouttotry)
exten = _XX,n,Dial,SIP/xlite1
exten = _XX,n,HangUp


When call is placed, the following debug info is shown, after the last
line, it stalls until caller gives up:


INFO_IND ID=001 #0x040a LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x7e
  InfoElement = 04

CAPI: no interface for PLCI = 0x101 MN = 0x40a
INFO_RESP ID=001 #0x040a LEN=0012
  Controller/PLCI/NCCI= 0x101

CAPI: INFO_IND no interface for PLCI=0x101
INFO_IND ID=001 #0x040b LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = 89

CAPI: no interface for PLCI = 0x101 MN = 0x40b
INFO_RESP ID=001 #0x040b LEN=0012
  Controller/PLCI/NCCI= 0x101

CAPI: INFO_IND no interface for PLCI=0x101
CONNECT_IND ID=001 #0x040c LEN=0038
  Controller/PLCI/NCCI= 0x101
  CIPValue= 0x10
  CalledPartyNumber   = 8161
  CallingPartyNumber  = 09 8017
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a3
  LLC = default
  HLC = 91 81
  AdditionalInfo
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- CONNECT_IND (PLCI=0x101,DID=61,CID=17,CIP=0x10,CONTROLLER=0x1)
hfcpci: msn='*' DNID='61' MSN
  == hfcpci: Incoming call '17' - '61'

-- 
Konrads Smelkovs
Applied IT sorcery.
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[Asterisk-Users] TDM Card FXO Question

2005-09-05 Thread Robert Webb

I have a TDM card with one FXO and one FXS. I am trying to make sure I
understand correctly the TX and RX Gain in the Zapata.conf correctly. If
I have a phone cord plugged into an FXO port tied into a POTS line and
boost the TXGain, am I correct in thinking that the audio going back to
the phone company is boosted by X percentage??

TIA,
Robert



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[Asterisk-Users] Re: ztdummy and Linux 2.6.13-rc7

In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
 
 Thanks for the input Tony, but the instructions that Rob Thomas wrote 
 took care of my issue.

That's good news. I saw what Rob wrote, and hadn't been aware of HPET
before, so I was glad to find out.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] TDM Card FXO Question

2005-09-05 Thread Rich Adamson


 I have a TDM card with one FXO and one FXS. I am trying to make sure I
 understand correctly the TX and RX Gain in the Zapata.conf correctly. If
 I have a phone cord plugged into an FXO port tied into a POTS line and
 boost the TXGain, am I correct in thinking that the audio going back to
 the phone company is boosted by X percentage??

Yes, but its not really a percentage. The number that is entereed
into txgain is a db number (of gain or loss). So, txgain=5 says the
audio transmitted to the pstn is 5 db greater then the default value
of 0 db.


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Re: [Asterisk-Users] asterisk CAPI dial-in issues

On Mon, 5 Sep 2005, Konrads Smelkovs wrote:
 Hello configuration as follows, dial-out works:
 
 capi.conf:
 [hfcpci]
 ;;PointToPoint (55512-0)
 isdnmode=MSN
 incomingmsn=*
 ;msn=61
 controller=1
 devices=2
 context=incoming
 
 extensions.conf:
 [incoming]
 exten = _XX,1,Playback(demo-abouttotry)
 exten = _XX,n,Dial,SIP/xlite1
 exten = _XX,n,HangUp
 
 
 When call is placed, the following debug info is shown, after the last
 line, it stalls until caller gives up:
 
 
 INFO_IND ID=001 #0x040a LEN=0016
   Controller/PLCI/NCCI= 0x101
   InfoNumber  = 0x7e
   InfoElement = 04
 
 CAPI: no interface for PLCI = 0x101 MN = 0x40a
 INFO_RESP ID=001 #0x040a LEN=0012
   Controller/PLCI/NCCI= 0x101
 
 CAPI: INFO_IND no interface for PLCI=0x101
 INFO_IND ID=001 #0x040b LEN=0016
   Controller/PLCI/NCCI= 0x101
   InfoNumber  = 0x18
   InfoElement = 89
 
 CAPI: no interface for PLCI = 0x101 MN = 0x40b
 INFO_RESP ID=001 #0x040b LEN=0012
   Controller/PLCI/NCCI= 0x101
 
 CAPI: INFO_IND no interface for PLCI=0x101
 CONNECT_IND ID=001 #0x040c LEN=0038
   Controller/PLCI/NCCI= 0x101
   CIPValue= 0x10
   CalledPartyNumber   = 8161
   CallingPartyNumber  = 09 8017
   CalledPartySubaddress   = default
   CallingPartySubaddress  = default
   BC  = 80 90 a3
   LLC = default
   HLC = 91 81
   AdditionalInfo
BChannelinformation= default
Keypadfacility = default
Useruserdata   = default
Facilitydataarray  = default
 
 -- CONNECT_IND (PLCI=0x101,DID=61,CID=17,CIP=0x10,CONTROLLER=0x1)
 hfcpci: msn='*' DNID='61' MSN
   == hfcpci: Incoming call '17' - '61'

There are no more messages?
SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not 
signal the call to Asterisk.

What card/driver do you use? And what kind of line is it (ptmp of a pbx)?

Armin

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Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration

2005-09-05 Thread Matt

Wel I tried by hand first that failed.. so I emptied the table..
and tried the perl script.   That gave errors of category can not be
NULL.   and didn't insert anything into the table.
If I allowed NULLS for category it put things in pretty much exactly
how I put them in...

MySQL RealTime Static seems to see the settings as it goes through and
does the select.. but the it just kinda ignores them.. in that it says
things like:

Message Review disabled globaly.

When I have something like
review = yes
(or whatever the statement is to allow reviewing).

On 9/4/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 How did you convert your voicemail.conf file into RT Static? Did you use the
 perl script?
 
 -Matthew
 
 
  From: Matt [EMAIL PROTECTED]
  Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Date: Sun, 4 Sep 2005 20:37:34 -0400
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
 
  I should add to this... I understand to make the table.. but when I
  make it.. asterisk selects it but seems to ignore things.   No where
  have I found documented what the var_category and such are... what
  numbers do I put in there?!?!
 
  On 9/4/05, Matt [EMAIL PROTECTED] wrote:
  Hi,
  When using asterisk real-time with mysql voicemail integration...
  where exactly do I put the options like the [PBX] tag, and how long
  silence can be, etc?
 
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Re: [Asterisk-Users] asterisk CAPI dial-in issues

2005-09-05 Thread Sergio Chersovani


Armin Schindler ha scritto:


There are no more messages?
SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not 
signal the call to Asterisk.


 

The sending complete field is pretty new in the libcapi, maybe he just 
need to update the capi20 lib.


Sergio
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[Asterisk-Users] Re:initiate call with asterisk - authentication error telnetting to Manager API

2005-09-05 Thread Eric

Hi

I have the following error in the logs when trying to login to
the Manager API:

 ... tried to authenticate with non-existant user 'admin'

Login is as follows:

 Action: Login
 Username: admin
 Secret: secret
 Action: Originate
 Channel: SIP/snom
 Context: default
 Exten: 2412
 Priority: 1
 Callerid: Asterisk Automatic Wardial
 Action: Logoff


=  managet.conf is === 
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

;[mark]
user=admin
secret=secret
;deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0

read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
==

What gives?

Thanks

Eric 


Adam Dobrin said:
 http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
 
 Eric wrote:
 
 I would like to initiate a call in asterisk (say with cron)
 so that this call rings on the destination number _and_
 on an asterisk extension.
 
 How would I achieve this?
 
 thanks
 
  
 
 
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-- 
Eric Smith
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[Asterisk-Users] A good HW

2005-09-05 Thread Jan Buchal

Hello,

I am working in non profit organisation Brailcom which develop Free
Software for blind and visually impaired people. Now we think about a
new switchboard for our current work and for better communication with
our blind clients. If I good understand can be useful asterisk with
some hw card for us.

We have this requests:

- linking with current analog provider

- forwarding usually phone service to voip

- forwarding voip to usually phone service

- we will to use in Czech republic, Europe

If I good understand for this we can use some HW which list I founded in
asterisk documentation. However I do not know what kind will be good for
us. The price is very important for us of course. Do you help me please
and suggest some telephony card?

thanks.



-- 

Jan Buchal
Tel: (00420) 224921679
Mob: (00420) 608023021

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[Asterisk-Users] Re: initiate call with asterisk - authentication error telnetting to Manager API

In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
 Hi
 
 I have the following error in the logs when trying to login to
 the Manager API:
 
  ... tried to authenticate with non-existant user 'admin'
 
 =  managet.conf is === 

I assume you mean manager.conf :-)

 ; Asterisk Call Management support
 ;
 [general]
 enabled = yes
 port = 5038
 bindaddr = 0.0.0.0

You need to change this:
 ;[mark]
 user=admin

to this:
[admin]

 secret=secret
 ;deny=0.0.0.0/0.0.0.0
 permit=127.0.0.1/255.255.255.0
 
 read = system,call,log,verbose,command,agent,user
 write = system,call,log,verbose,command,agent,user
 ==

The user name goes in the brackets, not after a user=.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Re: initiate call with asterisk - authentication error telnetting to Manager API

2005-09-05 Thread Eric

Thanks for the help.

Ok, it is now authenticating.

But the command:
 Channel: SIP/snom
 Context: default
 Exten: 2412

causes no action and nothing in the logs.

Any idea?

Thanks a lot

Eric
Tony Mountifield said:
 In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
  Hi
  
  I have the following error in the logs when trying to login to
  the Manager API:
  
   ... tried to authenticate with non-existant user 'admin'
  
  =  managet.conf is === 
 
 I assume you mean manager.conf :-)
 
  ; Asterisk Call Management support
  ;
  [general]
  enabled = yes
  port = 5038
  bindaddr = 0.0.0.0
 
 You need to change this:
  ;[mark]
  user=admin
 
 to this:
 [admin]
 
  secret=secret
  ;deny=0.0.0.0/0.0.0.0
  permit=127.0.0.1/255.255.255.0
  
  read = system,call,log,verbose,command,agent,user
  write = system,call,log,verbose,command,agent,user
  ==
 
 The user name goes in the brackets, not after a user=.
 
 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] asterisk CAPI dial-in issues

On Mon, 5 Sep 2005, Sergio Chersovani wrote:
 Armin Schindler ha scritto:
 
  There are no more messages?
  SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
  signal the call to Asterisk.
  
  
  
 The sending complete field is pretty new in the libcapi, maybe he just need to
 update the capi20 lib.

Yes, that field is new in the lib. But chan_capi does not use this field, it 
waits for the INFO_IND for that IE.

Armin

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Re: [Asterisk-Users] A few questions before final proposal...

2005-09-05 Thread Mark Phillips


Hi Kurth,

I'm in NJ. I'd be happy to help you out either on the phone or in person.

Gimme a call 973 828 1625

Mark

Kurth Bemis wrote:

Adam,

Thanks for your help.

Does anyone know or is anyone an * guru in the New Hampshire/Vermont area?

how about this example.

User1 sits at his desk, a call comes in.(doesn’t matter how the call 
gets to his phone, DID or exten) he needs to go into the warehouse to 
look at something.  He places the call on hold, notes the line and goes 
to the warehouse.  Once there, he picks up another handset, presses the 
button for the line he would like to pickup.  How is this done with 
FOP?  Everyone has access to FOP, not just the system operator?  Would 
the user be better off transferring the call to that phone in the 
warehouse?


How have others implemented this feature?

~kurth

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Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration

2005-09-05 Thread Matthew Boehm

 MySQL RealTime Static seems to see the settings as it goes through and
 does the select.. but the it just kinda ignores them

Strange. Have you verified this behavior with ODBC RealTime? The code
that parses the results is virtually identical so I don't see this as a
mysql-rt specific issue.

-Matthew


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Re: [Asterisk-Users] asterisk CAPI dial-in issues

It is connected to the PBX, alcatel omnipcx.
My libcapi20is dated Oct 21, 2004. 
Where can I get the libcapi? There seems to be 100 sources and none
smells official.


On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote:
 Armin Schindler ha scritto:
 
 There are no more messages?
 SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
 signal the call to Asterisk.
 
 
 
 The sending complete field is pretty new in the libcapi, maybe he just
 need to update the capi20 lib.
 
 Sergio
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 22

2005-09-05 Thread Nguyen Trung Tin

Hi All

I have problem with LIBMFCR2 for once Exchange

I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2.
my system:

 Asterisk CVS 1.1.X
LibMFCR2 pre.005 and unicall-pre.005

this's my setting

and my setting wanpipe1.conf
## WANPIPE1 Configuration File### Date: Mon Sep 5 15:37:16 GMT+7 2005## Note: This file was generated automatically# by /usr/sbin/wancfg program.## If you want to edit this file, it is# recommended that you use wancfg program# to do so.## Sangoma Technologies Inc.#
[devices]wanpipe1 = WAN_AFT, Comment
[interfaces]w1g1 = wanpipe1, , TDM_VOICE, Comment
[wanpipe1]CARD_TYPE = AFTS514CPU = ACommPort = PRIAUTO_PCISLOT = YESPCISLOT = 2PCIBUS = 1FE_MEDIA= E1FE_LCODE= HDB3FE_FRAME= NCRC4FE_LINE= 1TE_CLOCK = NORMALTE_REF_CLOCK = 0ACTIVE_CH= ALLTE_HIGHIMPEDANCE= YESFE_TXTRISTATE= NOMTU = 2100UDPPORT = 9000TTL= 255IGNORE_FRONT_END = NOTDMV_SPAN= 1TDMV_DCHAN= 0
[w1g1]ACTIVE_CH= ALLTDMV_ECHO_OFF= NO 

and my setting /etc/zaptel.conf
## Zaptel Configuration File## This file is parsed by the Zaptel Configurator, ztcfg### First come the span definitions, in the format# span=span num,timing,line build out (LBO),framing,coding[,yellow]# # The timing parameter determines the selection of primary, secondary, and# so on sync sources. If this span should be considered a primary sync# source, then give it a value of "1". For a secondary, use "2", and so on.# To not use this as a sync source, just use "0"## The line build-out (or LBO) is an integer, from the following table:# 0: 0 db (CSU) / 0-133 feet (DSX-1)# 1: 133-266 feet (DSX-1)# 2: 266-399 feet (DSX-1)# 3: 399-533 feet (DSX-1)# 4: 533-655 feet (DSX-1)# 5: -7.5db (CSU)# 6: -15db (CSU)# 7: -22.5db (CSU)## The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1## Note: "d4" could be referred
  to as
 "sf" or "superframe" ## The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1## E1's may have the additional keyword "crc4" to enable CRC4 checking## If the keyword "yellow" follows, yellow alarm is transmitted when no# channels are open.##span=1,0,0,esf,b8zs#span=2,1,0,esf,b8zs#span=3,0,0,ccs,hdb3,crc4
#span=1,0,0,cas,hdb3,crc4#span=1,0,0,cas,hdb3,crc4,yellow
#span=1,0,0,cas,hdb3,crc4#span=1,0,0,cas,hdb3,ncrc4
#canthospan=1,2,0,cas,hdb3#span=1,1,0,cas,hdb3#span=1,1,0,ccs,hdb3
#span=1,0,0,ccs,hdb3,yellow
#span=2,0,0,cas,hdb3,crc4#span=3,0,0,cas,hdb3,crc4#
#cas=1-15:1001#cas=17-31:1001

#cas=1-15:0101#cas=17-31:0101
#cas=1-15:#cas=17-31:
#cas=1-15:1101#cas=17-31:1101
cas=1-15:1101cas=17-31:1101
#bchan=1-15#bchan=17-31
dchan=16
#alaw=1-31alaw=1-15alaw=17-31
# Next come the dynamic span definitions, in the form:# dynamic=driver,address,numchans,timing#
#dynamic=w1g1,w1g1/16,31,0
# Where driver is the name of the driver (e.g. eth), address is the# driver specific address (like a MAC for eth), numchans is the number# of channels, and timing is a timing priority, like for a normal span.# use "0" to not use this as a timing source, or prioritize them as# primary, secondard, etc. Note that you MUST have a REAL zaptel device# if you are not using external timing.## dynamic=eth,eth0/00:02:b3:35:43:9c,24,0## Next come the definitions for using the channels. The format is:# device=channel list## Valid devices are:## "em" : Channel(s) are signalled using EM signalling (specific# implementation, such as Immediate, Wink, or Feature Group D# are handled by the u
 serspace
 library).# "fxsls" : Channel(s) are signalled using FXS Loopstart protocol.# "fxsgs" : Channel(s) are signalled using FXS Groundstart protocol.# "fxsks" : Channel(s) are signalled using FXS Koolstart protocol.# "fxols" : Channel(s) are signalled using FXO Loopstart protocol.# "fxogs" : Channel(s) are signalled using FXO Groundstart protocol.# "fxoks" : Channel(s) are signalled using FXO Koolstart protocol.# "sf" : Channel(s) are signalled using in-band single freq tone.#Syntax as follows: # channel# = sf:rxfreq,rxbw,rxflag,txfreq,txlevel,txflag#rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)#bandwith in hz (typically 10.0), rxflag is either 'normal' or#'inverted', txfreq is tx tone freq in hz, txlevel is tx tone
 #level in dbm, txflag is either 'normal' or 'inverted'. Set #rxfreq or txfreq to 0.0 if that tone is not desired.# "unused" : No signalling is performed, each channel in the list remains idle# "clear" : Channel(s) are bundled into a single span. No conversion or# signalling is performed, and raw data is available on the master.# "indclear": Like "clear" except all channels are treated individually and# are not bundled. "bchan" is an alias for this.# "rawhdlc" : The zaptel driver performs HDLC encoding and

Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration

2005-09-05 Thread Matt

I've not yet tried ODBC.
And if I do the select statement that it's doing I get back the
results it wants.

Does anyone have documentation on the fields?

 `id` - Assume just a key ID.
 `cat_metric` -- ?
 `var_metric` -- ?
 `commented` -- assume if the value is commented ; or not.
 `filename` -- the filename ie voicemail.conf
 `category` -- ?
 `var_name` -- the variable name
 `var_val` -- variable value.

Is it possible it's not reading the config because of an incorrect var
or cat_metric?  But if so what are they suppose to read?

On 9/5/05, Matthew Boehm [EMAIL PROTECTED] wrote:
  MySQL RealTime Static seems to see the settings as it goes through and
  does the select.. but the it just kinda ignores them
 
 Strange. Have you verified this behavior with ODBC RealTime? The code
 that parses the results is virtually identical so I don't see this as a
 mysql-rt specific issue.
 
 -Matthew
 
 
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[Asterisk-Users] compiling asterisk

2005-09-05 Thread Dante Renda

I am getting an error compiling latest stable version from CVS, but compiling 
CVS-HEAD on the same machine compile ok.
I have installed TE110P

the error is

chan_zap.c: In function `zt_handle_event':
chan_zap.c:2772: error: `ZT_EVENT_DTMFDIGIT' undeclared (first use in this 
function)
chan_zap.c:2772: error: (Each undeclared identifier is reported only once
chan_zap.c:2772: error: for each function it appears in.)
chan_zap.c: In function `load_module':
chan_zap.c:7700: warning: passing arg 1 of `pri_set_error' from incompatible 
pointer type
chan_zap.c:7701: warning: passing arg 1 of `pri_set_message' from incompatible 
pointer type
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1



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Re: [Asterisk-Users] unicall deploy

2005-09-05 Thread acriollo

Thanks Guillermo.
Can you share your experience with the software ?
Network Architecture, Linux Kernel, etc. ANI, DNIS, etc
And very important, version of the unicall library, and if you had any
problems receiving and making calls.

How many calls do you have to outside ?

Can you shara with us your config files for the unicall ?

Regards. Saludos de Mexico



2005/9/3, Guillermo Freige [EMAIL PROTECTED]:
 I´m using an unicall box with 4 E1 lines getting between 6000-15000 calls
 per day, and between 15-30 operators using AgentLogin, all using R2
 signaling to the telco and a local PBX. I´m using the Argentina variant, and
 using the last version of unicall 0.0.2 and asterisk 1.0.7
 
 Guillermo
 
 
 From: acriollo [EMAIL PROTECTED]
 Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] unicall deploy
 Date: Sat, 3 Sep 2005 15:04:20 -0500
 
 Hi every one .
 
 There are any out there that have a unicall deploy working without problem
 ?
 Can give me some tips or referenece about his config ?
 
 Regards
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Re: [Asterisk-Users] sending fax

2005-09-05 Thread Chris Shipman

I've  seen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.
It would need to be able to print to as local printer and then send out
Asterisk.

Chris

- Original Message - 
From: Arne Morten Johansen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 05, 2005 6:27 AM
Subject: SV: [Asterisk-Users] sending fax


 What about faxing yourself if you don't have a scanner?

 -Opprinnelig melding-
 Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Johan van
Tongeren
 Sendt: 5. september 2005 09:11
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: RE: [Asterisk-Users] sending fax

 [macro-fax-dialing]
 exten = s,1,SetCIDNum(0${CALLERIDNUM})
 exten = s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t)
 exten = s,3,Goto(900)
 exten = s,103,Goto(900)
 exten = s,900,Busy
 exten = s,901,Hangup

 -Oorspronkelijk bericht-
 Van: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Namens Chris Shipman
 Verzonden: maandag 5 september 2005 7:22
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [Asterisk-Users] sending fax

 I've read alot on the wiki about sending and receiving faxes thru
 asterisk.
 I've gotten the receive to work great.My question is how does one
 send a
 fax?
 I see lots of instructions about how to send the image to asterisk by
 email,
 etc.  The problem is how does  one make the image of the fax to
 begin
 with?   Has anyone come up with a good solution for this?


 Regards,


 Chris


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[Asterisk-Users] ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)

Hello,

I have the following setup:

(*)---IP---Micronet 5012 H.323 box --- POTS --- PBX (Alcatel OmniPCX)
Grand idea is to use the micronet's POTS interfaces to connect SIP
phones to the PBX and to the PSTN. I think i even managed my way in
the arcane and cryptic management interface of that appliance, but I
am stuck against theese messages:

-- Executing Dial(SIP/xlite1-7a03, H323/120/smallbox) in new stack
---   h323_request - data 120/smallbox format 0x4 (ulaw)
---   find_peer
+++   find_peer
+++   h323_request
---   h323_call- 120/smallbox
+++   h323_call
-- Called 120/smallbox
---   onNewCallCreated ooh323c_1
---   find_call
+++   find_call
 Outgoing call smallbox(ooh323c_1) - Codec prefs - (gsm|alaw|ulaw)
 Adding capabilities to call(outgoing, ooh323c_1)
 Adding gsm capability to call(outgoing, ooh323c_1)
 Adding g711 alaw capability to call(outgoing, ooh323c_1)
 Adding g711 ulaw capability to call(outgoing, ooh323c_1)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_1
---   setup_rtp_connection
---   find_call
+++   find_call
+++   setup_rtp_connection
--- onAlerting ooh323c_1
---   find_call
+++   find_call
+++ onAlerting ooh323c_1
 -- H323/smallbox-f14a is ringing
---   onCallEstablished ooh323c_1
---   find_call
+++   find_call
+++   onCallEstablished ooh323c_1
 -- H323/smallbox-f14a answered SIP/xlite1-7a03
 -- Attempting native bridge of SIP/xlite1-7a03 and H323/smallbox-f14a
---   h323_set_peer - H323/smallbox-f14a
Sep  5 18:28:27 NOTICE[27211]: src/chan_h323.c:2749
h323_convertAsteriskCapToH323Cap: Don't know how to deal with mode
0x40 (slin)
---   close_rtp_connection
---   find_call
+++   find_call
+++   close_rtp_connection
---   onCallCleared ooh323c_1
---   find_call
+++   find_call
---   h323_hangup
 hanging smallbox
+++   h323_hangup
 == Spawn extension (default, 120, 1) exited non-zero on 'SIP/xlite1-7a03'
---   h323_destroy
 Destroying smallbox
+++   h323_destroy


I think that, if it would not try to do native bridge, but transcode
the sound, it would work.
Perhaps there is an option, like forcetranscode?
-- 
Konrads Smelkovs
Applied IT sorcery.
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RE: [Asterisk-Users] No DID on ZAP

2005-09-05 Thread Alexander Lopez

 How is your line provisioned?? (EW, PRI, Trunks, etc.)



 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Darren Wright
 Sent: Monday, September 05, 2005 5:36 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] No DID on ZAP 
 
 I can't seem to get any ZAP trunks on my TE110P to match any 
 extensions for incoming DID.
 
 
 I've even used the exten = _X.,1And it still will not match that.
 All I get is:
 
 
  -- Starting simple switch on 'Zap/1-1'
   == Starting Zap/1-1 at zap-custom,s,1 failed so falling 
 back to exten 's'
   == Starting Zap/1-1 at zap-custom,s,1 still failed so 
 falling back to context 'default'
 
 The only think it will match is exten = s,1
 
 And then it works fine...all Callerid is perfect.
 
 Any ideas?
 
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Re: [Asterisk-Users] GotoIf sample...

2005-09-05 Thread anderson


exten = ,1,GotoIf($[${CALLERIDNUM} = 2000]?3)
exten = ,2,GotoIf($[${CALLERIDNUM} = 2001]?4:5)
exten = ,3,WaitExten(10)
exten = ,4,WaitMusicOnHold(60)
exten = ,5,Hangup()

If the caller ID of the caller is 2000 then run WaitExten(10)
if it's not 2000 and it's 2001 put the person on hold, if it's neither
Hangup.

Hope that helps.




On Mon Sep 05, 2005 at 03:19:20AM -0700, ryan nalupa wrote:
 hi everyone. can anyone provide me concrete examples on how to use the GotoIf 
 application? can't figure out how to use it in my dialplan coz im having 
 errorsthanks! : )
 
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RE: [Asterisk-Users] chan_unical-MFC/R2 CPU usage problem

2005-09-05 Thread Hadi Jadallah

Hi,

I was running asterisk using the init.d script. Turns out if I don’t use the 
init.d script and run asterisk either directly 'asterisk -vg ' or 
putting safe_asterisk in rc.local then the cpu utilization problem does not 
happen anymore.

 
 Hi,
 
 My variant is standard ITU, I tried almost all versions I 
 could put my hand on to no avail.
 I tried also to profile the channel and related libraries to 
 no avail as my profiling skills on linux are abit lacking.
 If anybody with this problem and knows how to profile 
 multithreaded apps on linux then we might at least pinpoint 
 the location of the error.
 I cant realy put the machine into active duty if I cant solve 
 the problem.
 
 Btw, what version of libtiff are you using? It difficult to 
 believe that it might be related as I don’t need the fax 
 functionality. Mine is the version that comes with CentOS 4.1 
 which is 3.6.1.
 
 Hadi.
 
 
  Message: 21
  Date: Wed, 24 Aug 2005 13:01:15 -0300
  From: Leonardo Gomes Figueira [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] chan_unical-MFC/R2 CPU usage problem
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Message-ID: [EMAIL PROTECTED]
  Content-Type: text/plain; charset=UTF-8; format=flowed
  
  Hi,
  
  Hadi Jadallah wrote:
   I have installed chan_unicall and MFC/R2 successfully, and 
  is runnign fine.
   But I noticed that once unicall is installed, asterisk CPU 
  usage as reported by 'top', jumps to 99% every few seconds.
   I have no incoming calls, and I have even removed the E1 
  lines from card and I tried almost everything possible but I 
  was not successful in determining the cause of this high cpu 
  utilization.
  
  It happens here too. But only when there is at least one 
  Unicall channel 
up. It does not happen on every call and I couldn't find a 
  pattern yet.
  
   My setup includes:
   asterisk 1.0.9, libpri 1.0.9, and zaptel 1.0.9.1
   Unicall 0.0.3pre3 and tried unicall-0.0.2c
   Digium TE410p
   Intel SE7520BD2 with Xeon 3.4GHz, 2 Gig Ram
  
  Almost the same setup here. The only difference is hardware. 
  Soyo + P4 
  2.8 512MB.
  
  You didn't specify your R2 variant. Here it's the brazilian and the 
  Asterisk box is connected on an Ericsson MD110.
  
  I'll upgrade to 0.0.3pre4 now. Maybe it's fixed in this version ?
  
  Bye,
  
  Leonardo
  
  
  
  
 
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[Asterisk-Users] more accounts

2005-09-05 Thread FaberK

Hi guys,
one question:
I've got 2 IAX accounts, and I would like to let use them in the same time, so that if one is busy I can call using the other?

Thanks-- .:FaberK:.
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[Asterisk-Users] User authentication and privileges

2005-09-05 Thread Mark Elkins

I want to authenticate a user before he is able to use the phone. I also
want to set his privilege as to where he is allowed to call to...

Preferably, the password should be their VoiceMail password,  (every
extension (or is that user?) can have voicemail defined - even if its
not in use?)

...one should be able to enter the password (variable length) as part of
the dial sequence - eg the number to call is 0113140077 and the password
is 1234 so dial something like *1234*0113140077 (no prompting!) and what
should be written to the Accounts file should rather be the extension
that that password is good for... (effectively - the User). 
This way, using voicemail.conf, users can manage their own passwords.

I've seen some wiki stuff on AGI's that allow one to glean for user
passwords..

If the system is smart (and the user not so), after dialing a trunk that
needs a password and none were provided - then asterisk can prompt for
it.

It would also be cool if certain extensions did not need a password...
(phone in MD's office?, Switchboard, Fax (maybe)) - this needs a flag
against the extension - which could be a Privilege Flag.

Privilege Flag: (suggestion)
0=internal calls (and emergency/911)
1=local calls
2=long distance
3=cellular
4=no barring at all (international)

(Somehow need to Tag the class (privilege level) that a number falls
into)

Then what about an additional field in the voicemail.conf file that
specifies what privilege a person has - ie from a phone with zero
privilege, a user with priv 4 can use his password to make an
international call...

I say user rather than extension because a user should be able to
call from any extension with their own password - the user has the
restriction - not the extension.

Anyone got anything like this?

-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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[Asterisk-Users] Asterisk Follow ME

2005-09-05 Thread Vladyslav

Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.

When call goes via IAX and calling part accept the call on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.

If anyone faced with such problem please let me know. I need to know
whether it's bug or just configuration issue.

 Thank U.



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[Asterisk-Users] Re: initiate call with asterisk - authentication error telnetting to Manager API

In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
 Thanks for the help.
 
 Ok, it is now authenticating.
 
 But the command:
  Channel: SIP/snom
  Context: default
  Exten: 2412
 
 causes no action and nothing in the logs.
 
 Any idea?

Well in your original post you had:

 Action: Login
 Username: admin
 Secret: secret
 Action: Originate
 Channel: SIP/snom
 Context: default
 Exten: 2412
 Priority: 1
 Callerid: Asterisk Automatic Wardial
 Action: Logoff

You need to terminate each command with a blank line, like this:

---start---
 Action: Login
 Username: admin
 Secret: secret

 Action: Originate
 Channel: SIP/snom
 Context: default
 Exten: 2412
 Priority: 1
 Callerid: Asterisk Automatic Wardial

 Action: Logoff

---end---

If you are already doing that, I'm not sure what your problem could be
without more information.

You should probably read the responses too, to check for errors.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] BT100 and BETA 1.0.7.11

2005-09-05 Thread Bartosz Jozwiak


Hi,

Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ?
For me so far no success.

Bartosz
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[Asterisk-Users] Provisioned, Down, Active, but D-channel seems to be fine

2005-09-05 Thread steve


Hi,

I'm seeing rather odd behaviour on a new box with TE110P card.

I'm running the TE110P span with ccs,hdb3,crc4 in pri_net, connected to a 
second machine with a TE410P in pri_cpe.

The span is idle.

I'm using pri intense debug span 1 and can see the RRs going back and 
forth.

So - things are running along with the span showing Provisioned, Up, 
Active in pri show span 1.

Something happens and the span logs an alarm, and goes down.  All the 
channels go down.

Almost immediately, the alarm clears.  I see the RRs again.

But, pri show span 1 shows the span now as Provisioned, Down, Active.

But all along we are exchanging RRs (receive ready) with the remote 
system.

So I'm pretty sure that the span is actually up again.  But why doesn't 
Asterisk agree...?

(Actually, I suspect that at the level of the D-channel it didn't really
go down, we just had a tiny layer-1 glitch).

Looking in the chan_zap code around line 8000, I see that when it gets an 
ALARM from zaptel is clears the DCHAN_UP flag.  But I don't understand the 
process by which the DCHAN_UP gets set to yes again.

Any comments or pointers?

Steve

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[Asterisk-Users] putty and winscp

2005-09-05 Thread Dean Collins









I found this great link, it has both putty and winscp
available for download (with asterisk these are invaluable tools).



http://www.cs.sunyit.edu/network/downloads/





It also has one of the last downloadable copies of PGP 8.1
that I know is available (still the best pgp program and able to be licensed
with fake serials) ost of the others on the net have been erased by the pgp
corp or lead to the new 9.0 version.





Cheers,

Dean










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Re: [Asterisk-Users] asterisk CAPI dial-in issues

Oh, and I am using chan_cap via mISDN on HFCPCI.  

On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote:
 It is connected to the PBX, alcatel omnipcx.
 My libcapi20is dated Oct 21, 2004.
 Where can I get the libcapi? There seems to be 100 sources and none
 smells official.
 
 
 On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote:
  Armin Schindler ha scritto:
 
  There are no more messages?
  SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
  signal the call to Asterisk.
  
  
  
  The sending complete field is pretty new in the libcapi, maybe he just
  need to update the capi20 lib.
 
  Sergio
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 --
 Konrads Smelkovs
 Applied IT sorcery.
 


-- 
Konrads Smelkovs
Applied IT sorcery.
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Re: [Asterisk-Users] Provisioned, Down, Active, but D-channel seems to be fine

2005-09-05 Thread Kevin P. Fleming


[EMAIL PROTECTED] wrote:


But, pri show span 1 shows the span now as Provisioned, Down, Active.

But all along we are exchanging RRs (receive ready) with the remote 
system.


The telco has turned your circuit 'administratively down' so their 
operator console would stop getting spammed with 'PRI down' errors.

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Re: [Asterisk-Users] Provisioned, Down, Active, but D-channel seems to be fine

2005-09-05 Thread steve



On Mon, 5 Sep 2005, Kevin P. Fleming wrote:

 [EMAIL PROTECTED] wrote:
 
  But, pri show span 1 shows the span now as Provisioned, Down, Active.
  
  But all along we are exchanging RRs (receive ready) with the remote 
  system.
 
 The telco has turned your circuit 'administratively down' so their 
 operator console would stop getting spammed with 'PRI down' errors.


Kevin:

Thanks for taking the trouble to actually read my post.

The other end of the circuit is another Asterisk box.

Steve

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Re: [Asterisk-Users] Provisioned, Down, Active, but D-channel seems to be fine

2005-09-05 Thread Kevin P. Fleming


[EMAIL PROTECTED] wrote:


Thanks for taking the trouble to actually read my post.


Doh! Blame it on my weekend laziness :-)


The other end of the circuit is another Asterisk box.


Hmm... I have never seen that happen before, Asterisk is pretty 
aggressive about bringing the D-channel up as soon as possible after the 
span itself is up.

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RE: [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina

2005-09-05 Thread Guillermo Freige


Carlos:
I have no problem. I was answering a question. :)
In fact I'm managing around 1 call/day. Argentina uses no ISDN standard 
by default, but the old R2 standard. My telco is Telefonica


Guillermo



From: Carlos Alperin [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Argentina - zapata.conf switchtype for 
Argentina

Date: Sun, 4 Sep 2005 21:48:27 -0400

Guillermo,

Switchtype depends on to which kind of PSTN are you connected to.

Are you connected to Telecom or Telefonica?, using PRI or FXO/FXS lines?

Normally both follows European Standards for Telephony (CCITT), not Bell
standars.

And in the case of Telecom they have a lot of Telettra equipment installed.

I hope this can help you.

Carlos Alperin
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Freige
Sent: Sunday, September 04, 2005 12:03 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Argentina - zapata.conf switchtype for
Argentina

Probably you need to use unicall+mfcr2 support instead of zapata, as
Argentina uses R2.

Guillermo


From: Leandro Rzezak [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Argentina - zapata.conf switchtype for 
Argentina

Date: Sat, 3 Sep 2005 18:54:59 -0300

Just to receive a recommendation on switchtype for Argentina, Buenos 
Aires,

114816.
  Thanks a lot

--
Leandro Rzezak
[EMAIL PROTECTED]


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Re: [Asterisk-Users] asterisk CAPI dial-in issues

On Mon, 5 Sep 2005, Konrads Smelkovs wrote:
 It is connected to the PBX, alcatel omnipcx.
 My libcapi20is dated Oct 21, 2004. 
 Where can I get the libcapi? There seems to be 100 sources and none
 smells official.

You can get the sources from isdn4linux.de via CVS:
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev login
  password: readonly
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev co isdn4k-utils/capi20
 
Armin

 On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote:
  Armin Schindler ha scritto:
  
  There are no more messages?
  SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not
  signal the call to Asterisk.
  
  
  
  The sending complete field is pretty new in the libcapi, maybe he just
  need to update the capi20 lib.
  
  Sergio
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 -- 
 Konrads Smelkovs
 Applied IT sorcery.
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Re: [Asterisk-Users] unicall deploy

2005-09-05 Thread Guillermo Freige


Hi.
I'm using kernel 2.4.27 over sarge, with 0.0.2c. I'm using Argentina variant 
of R2, and have no problems receiving or sending ANI with the telco. 99% of 
the calls are incoming ones, but I have a small percentaje of outgoing ones 
too. Using 0.0.2c I resolved all the problems I had with previous versions 
regarding occational 99% CPU loops and some protocol errors too. No problems 
receiving or sending calls now, but sending is much less tested.


unicall.conf

[channels]
language=es
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0

callgroup=1
pickupgroup=1

protocolclass=mfcr2
protocolvariant=ar,16,16
protocolend=co
group = 1
context=pbx
callerid=asreceived
channel = 1-15
;skip time slot 16
channel = 17-31
channel = 32-46
;skip time slot 47
channel = 48-62

protocolclass=mfcr2
protocolvariant=ar,16,4
protocolend=co
group = 2
context=telco412
channel = 63-77
;skip time slot 78
channel = 79-93
channel = 94-108
;skip time slot 109
channel = 110-124

Guillermo


From: acriollo [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] unicall deploy
Date: Mon, 5 Sep 2005 10:14:09 -0500

Thanks Guillermo.
Can you share your experience with the software ?
Network Architecture, Linux Kernel, etc. ANI, DNIS, etc
And very important, version of the unicall library, and if you had any
problems receiving and making calls.

How many calls do you have to outside ?

Can you shara with us your config files for the unicall ?

Regards. Saludos de Mexico



2005/9/3, Guillermo Freige [EMAIL PROTECTED]:
 I´m using an unicall box with 4 E1 lines getting between 6000-15000 
calls

 per day, and between 15-30 operators using AgentLogin, all using R2
 signaling to the telco and a local PBX. I´m using the Argentina variant, 
and

 using the last version of unicall 0.0.2 and asterisk 1.0.7

 Guillermo


 From: acriollo [EMAIL PROTECTED]
 Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] unicall deploy
 Date: Sat, 3 Sep 2005 15:04:20 -0500
 
 Hi every one .
 
 There are any out there that have a unicall deploy working without 
problem

 ?
 Can give me some tips or referenece about his config ?
 
 Regards
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RE: [Asterisk-Users] BT100 and BETA 1.0.7.11

2005-09-05 Thread Santiago Vega

Yes I did with no problems...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Monday, September 05, 2005 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11

Hi,

Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ?
For me so far no success.

Bartosz
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[Asterisk-Users] callerid...

2005-09-05 Thread Santiago Vega









Hi, asterisk Users, sorry for my bad English 

im really newbie with this excellent pbx. But I ve a
problem with callerid num when I recive a call from PSTN.



PSTN- SipGateWay(Welltech3504)- Asterisk- BT100

How can I configure my asterisk to receive the callerid from
callers and not the callerid from the extension of the SipGAteway



Extension of Gateway (sip.conf)

[115]

type=friend
; either friend (peer+user), peer or user

context=sip

user=115

host=dynamic

canreinvite=no

nat=no
; there is not NAT between phone and Asterisk

disallow=all
; need to disallow=all before we can use allow=

allow=ulaw
; Note: In user sections the order of codecs

allow=alaw








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Re: [Asterisk-Users] BT100 and BETA 1.0.7.11

2005-09-05 Thread Bartosz Jozwiak


I am missing some files my grandstream phone wants to download:
bootloader.bin. I cannot find that file in release 1.0.7.11.
Any ideas ?

Bartosz

- Original Message - 
From: Santiago Vega [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Thursday, August 25, 2005 4:24 PM
Subject: RE: [Asterisk-Users] BT100 and BETA 1.0.7.11



Yes I did with no problems...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Monday, September 05, 2005 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11

Hi,

Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ?
For me so far no success.

Bartosz
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Re: [Asterisk-Users] BT100 and BETA 1.0.7.11

2005-09-05 Thread Bartosz Jozwiak

It suppose to be bootload.bin . not bootloader.bin like in my previous 
mail.




I am missing some files my grandstream phone wants to download:
bootloader.bin. I cannot find that file in release 1.0.7.11.
Any ideas ?

Bartosz

- Original Message - 
From: Santiago Vega [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Thursday, August 25, 2005 4:24 PM
Subject: RE: [Asterisk-Users] BT100 and BETA 1.0.7.11



Yes I did with no problems...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Monday, September 05, 2005 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11

Hi,

Did anybody successfully updated Grandstream BT100 with BETA 1.0.7.11 ?
For me so far no success.

Bartosz
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[Asterisk-Users] Unexpected results with While and EndWhile applications

2005-09-05 Thread John Todd



I seem to be having a conceptual problem with the While and 
EndWhile applications.  It seems that on the first cycle, even if 
the result of the While is false that the enclosed applications 
will get run.  Is this expected?  It seems to be counter-intuitive, 
but I don't know what the intent of the While routines is.  I could 
of course put a GotoIf before the While loop to check to ensure 
that the first expression is true before entry into the While loop, 
but that seems redundant and ugly since the while point of While and 
EndWhile is to avoid the inelegance of GotoIf, I thought.


If anyone can't come up with a better explanation, I'll open a ticket 
on this but I'd like to first make sure that this behavior is not 
expected.



exten = 2231,1,Set(staticnumber=0)
exten = 2231,n,Set(counter=1)
exten = 2231,n,While($[${counter}${staticnumber}])
exten = 2231,n,NoOp(This part of the code should never run!)
exten = 2231,n,Set(counter=$[${counter}+1])
exten = 2231,n,EndWhile
exten = 2231,n,NoOp(This part of the code should be the only thing 
that gets run!)



Console output from dialing 2231:

-- Executing Set(SIP/2203-c134, staticnumber=0) in new stack
-- Executing Set(SIP/2203-c134, counter=1) in new stack
-- Executing While(SIP/2203-c134, 0) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should 
never run!) in new stack

-- Executing Set(SIP/2203-c134, counter=2) in new stack
-- Executing EndWhile(SIP/2203-c134, ) in new stack
-- Executing NoOp(SIP/2203-c134, This part of the code should 
be the only thing that gets run!) in new stack

*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux 
on 2005-09-03 23:27:34 UTC


JT
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[Asterisk-Users] Asterisk won't listen on another port

2005-09-05 Thread Aisling

Hello,

Hope somebody can help me Asterisk is behaving very
oddly and Im totally stumped! I have SER and Asterisk running on the
same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on
port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and
Asterisk will purely act as a voicemail server at the moment. However I cannot
get Asterisk to listen on a different port. It is my understanding that I just
need to set the port in sip.conf (port=5062) but that doesnt seem to be
working. When I type sip show settings into the console, I see SIP Port: 5060 in
Global Settings. When I run netstat tunap I
see:

x.x.x.x:5060
LISTEN

ser

127.0.0.1:5060 LISTEN

ser

0.0.0.0:2000
LISTEN

asterisk

0.0.0.0
:2727

asterisk

0.0.0.0:4520

asterisk

0.0.00:5060

asterisk

x.x.x.x:5060

ser

127.0.0.1:5060

ser

My config is like follows

;sip.conf

[general]

context=default

port=5062

bindaddr=0.0.0.0

srvlookup=yes

canreinvite=no

autocreatepeer=yes

[2092]

type=friend

username=2092

canreinvite=no

context=default

mailbox=2092

host=dynamic

nat=no dtmfmode=info

disallow=all

allow=ulaw

allow=alaw

;extensions.conf

;leave voice
messages

exten = 2092,
1, Voicemail(u2092)

exten = 2092,
2, Hangup

;play voice
messages

exten = ,
1, VoiceMailMain, s2092

;voicemail.conf

2092 = 2092, 2092, emailaddress

At the moment when a user dials to access voicemail,
ser forwards to x.x.x.x:5062 and with my current config (port 5062, bindaddr=0.0.0.0) nothing reaches asterisk. However when I
change this to (port=5062, bindaddr=x.x.x.x)the same address as ser, the phones start
registering with asterisk even though theyre configured to register with
port 5060 only! Basically I think Asterisk is still listening on 5060 and I
cant change it. I originally thought maybe I had multiple sip.confs on my machine but
when I do sip reload in the asterisk console, it says parsing
/etc/asterisk/sip.conf, so its definitely the correct file.

Do I need to change the asterisk port somewhere other that
sip.conf? Does anyone have other suggestions for what could be making Asterisk
behave so oddly?

Many thanks,

Aisling.

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RE: [Asterisk-Users] BT100 and BETA 1.0.7.11

2005-09-05 Thread Santiago Vega









Sorry , 

I only did the upgrade firmware version without erros!




 
  
  Software Version: 
  
  
   Program-- 1.0.7.11
  Bootloader-- 1.0.7.1 HTML-- 1.0.7.11
  VOC-- 1.0.1.0 
  
 






-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
Sent: Monday, September 05, 2005 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BT100 and BETA 1.0.7.11



I am missing some files my grandstream phone wants to download:

bootloader.bin. I cannot find that file in release 1.0.7.11.

Any ideas ?



Bartosz



- Original Message - 

From: Santiago Vega [EMAIL PROTECTED]

To: 'Asterisk Users Mailing List - Non-Commercial
Discussion' 

asterisk-users@lists.digium.com

Sent: Thursday, August 25, 2005 4:24 PM

Subject: RE: [Asterisk-Users] BT100 and BETA 1.0.7.11





 Yes I did with no problems...



 -Original Message-

 From: [EMAIL PROTECTED]

 [mailto:[EMAIL PROTECTED] On Behalf Of
Bartosz

 Jozwiak

 Sent: Monday, September 05, 2005 11:20 AM

 To: Asterisk Users Mailing List - Non-Commercial Discussion

 Subject: [Asterisk-Users] BT100 and BETA 1.0.7.11



 Hi,



 Did anybody successfully updated Grandstream BT100 with BETA
1.0.7.11 ?

 For me so far no success.



 Bartosz

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Re: [Asterisk-Users] Re: equipment configuration help

2005-09-05 Thread Erick Perez

I will as you suggested. And I will also post the configuration
somewhere so maybe some other newbie like me can benefit from it.

Millions of thanks for your help. If you ever travel to Panama in
central america let me know about it!!!

Thanks,

On 9/3/05, astgroups [EMAIL PROTECTED] wrote:
 Erick Perez wrote:
 
 So, with this i solve the issue on main office. But what about the two
 remote? they are so little that they will not let me place another *
 box there. The phones will be SIP and they are like this
 INTERNET--PIX--LAN(machines and sip phones). The pixes in those two
 offices have an ipsec tunnel with the main office via internet.
 I was thinking of placing the asterisk with a public IP so the remote
 phones can NAT outside to the public asterisk located in the main
 office.
 
 What do you think?
 
 On 9/2/05, asterisk groups [EMAIL PROTECTED] wrote:
 
 
 That is correct. Normally the layer 3 switches include advanced features
 such as QoS but they may be available on simpler layer 2 switches.
 
 I think the key words to look for are 'Managed, QoS (802.1p) with
 priority queues, VLAN, (802.1q)'...maybe even PoE if you go with some
 SIP phones in the future that can be powered by Power Over Ethernet.
 Something else to keep in mind.
 
 best of luck.
 
 On Thu, 2005-09-01 at 22:03 -0500, Erick Perez wrote:
 
 
 Why an L3? just for the QoS part?
 I checked the alliedtelesyn 8624T at $1000.00
 http://www.cdw.com/shop/products/default.aspx?EDC=772793
 
 but i also looked at the 8550T which has 48 port 10-100 but L2
 http://www.cdw.com/shop/products/default.aspx?EDC=773964RecommendedForEDC=772793RecoType=upsell
 at 900.00
 
 is the QoS different? sorry for the question but i keep reading that
 asterisk needs qos to function better.
 
 Thanks,
 
 On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote:
 
 
 Erick- Can't say if they will or not. In theory they should respect all
 outgoing traffic unless being filtered by another device such as your
 PIX. You might want to check with the ADSL router manufacturer just to
 be safe.
 
 
 On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote:
 
 
 Do i have to change the adsl routers? or just do QoS with the Layer 3 
 switches?
 Will my ADSL router respect the QoS setting when sending the packet to
 the Internet?
 
 
 On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote:
 
 
 Erick,
 
 After reviewing your original message a little closer it occurs to me
 that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400
 cards. These are Quad FXS or FXO cards that could receive the lines from
 your 8 analog line card.
 
 You'll still need an E1 card (Digium or Sangoma) to terminate your PRI
 line, but maybe with those TDM400 cards you can avoid the added cost of
 a channel bank.
 
 Regarding your WAN and branch offices;
 
 1. I've seen comments that tunneling VoIP traffic through IPSec can add
 overhead/delay that could impact voice quality. Something to keep in
 mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with
 IAX over the Internet not tunneled or encrypted and performance is fine.
 
 2. In your two locations with 15  50 users you should consider
 installing Asterisk boxes in those locations and trunking them together
 with IAX over the Internet. Perhaps go ahead and do the same thing with
 the smaller office. You can justify a small Asterisk implementation in
 an office with 5 phones.
 
 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth
 allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more
 economical D-Links. Put these behind your PIX. It is also recommended to
 do separate VLANs for any SIP hard phones you deploy. This adds another
 layer of security and reliability.
 
 Hope this helps.
 
 
 
 
 
 On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote:
 
 
 -M, The norstar has no E1 card, i will have to ask the nortel provider
 for the cost of it and configuration prices. I might end up paying the
 same as the channel bank.
 I was also thinking of using a Citel SIP-N-NORSTAR converter but its
 priced at around 3k. Too expensive because its only 24 ports and i
 have 32 nortel phones.
 
 According to this wiki
 http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel
 One problem with this approach is that in a Norstar system, it isn't
 easy to forward an extension to an outside line, which means Norstar
 phone users will have to remember to do something different when they
 want to call a user who has been switched to an IP phone for example.
 
 I guess that can be sorted out.
 
  Any manuals out there for configuration like
 [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1
 channel bank--- [Norstar]? (only the asterisk-t1-norstar part)
 
 Now another section, networking.
 The 3 offices are linked via VPNs like this
 Internet---ADSL Router-Cisco PIX  Firewall---LAN
 doin ip tunneling will solve all communication problems internally,
 but what about QoS and SIP phones being

[Asterisk-Users] res_features.so (Call Features Resource) not loading

2005-09-05 Thread Asterisk Sales

hello everybody,
i have updated my rpm asterisk to current cvs 1.0.9. I had been usingrpm asterisk which comeswith suse 9.2. the main reason i updated my asterisk to get the attended transfer feature. i haveinstalled anothercvs 
1.0.9 asterisk in Redhat 9 and it works perfect.
here what i found:

in suse 9.2:*CLI show modules

.
res_features.so Call Parking Resource Note: it shows CallParking Resource
.
..

in Redhat 9:
*CLI show modules.. 
res_features.so Call Features Resource Note: it shows Call Features Resource..
.

i think asterisk is not loading call features resources in suse 9.2 so i also added a line in
/etc/asterisk/modules.conf

load =res_features.so
.

still attended tranfer feature is not working. iwant to use suse 9.2 as it got IDSN driversupport. i use2 BRI cards.
please help

Thanks in advance
shaon (AU)
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Re: [Asterisk-Users] unicall deploy

2005-09-05 Thread acriollo

Thanks Guillermo.
Seems like nothing special is your configuration.
I have problems with outbound calls in a R2 Line here in mexico. I
dont now what is wrong yet.

Is goot to know that is working fine for you.

Regards.

2005/9/5, Guillermo Freige [EMAIL PROTECTED]:
 Hi.
 I'm using kernel 2.4.27 over sarge, with 0.0.2c. I'm using Argentina variant
 of R2, and have no problems receiving or sending ANI with the telco. 99% of
 the calls are incoming ones, but I have a small percentaje of outgoing ones
 too. Using 0.0.2c I resolved all the problems I had with previous versions
 regarding occational 99% CPU loops and some protocol errors too. No problems
 receiving or sending calls now, but sending is much less tested.
 
 unicall.conf
 
 [channels]
 language=es
 usecallerid=yes
 hidecallerid=no
 immediate=no
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=no
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 
 callgroup=1
 pickupgroup=1
 
 protocolclass=mfcr2
 protocolvariant=ar,16,16
 protocolend=co
 group = 1
 context=pbx
 callerid=asreceived
 channel = 1-15
 ;skip time slot 16
 channel = 17-31
 channel = 32-46
 ;skip time slot 47
 channel = 48-62
 
 protocolclass=mfcr2
 protocolvariant=ar,16,4
 protocolend=co
 group = 2
 context=telco412
 channel = 63-77
 ;skip time slot 78
 channel = 79-93
 channel = 94-108
 ;skip time slot 109
 channel = 110-124
 
 Guillermo
 
 From: acriollo [EMAIL PROTECTED]
 Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 To: Asterisk Users Mailing List - Non-Commercial
 Discussionasterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] unicall deploy
 Date: Mon, 5 Sep 2005 10:14:09 -0500
 
 Thanks Guillermo.
 Can you share your experience with the software ?
 Network Architecture, Linux Kernel, etc. ANI, DNIS, etc
 And very important, version of the unicall library, and if you had any
 problems receiving and making calls.
 
 How many calls do you have to outside ?
 
 Can you shara with us your config files for the unicall ?
 
 Regards. Saludos de Mexico
 
 
 
 2005/9/3, Guillermo Freige [EMAIL PROTECTED]:
   I´m using an unicall box with 4 E1 lines getting between 6000-15000
 calls
   per day, and between 15-30 operators using AgentLogin, all using R2
   signaling to the telco and a local PBX. I´m using the Argentina variant,
 and
   using the last version of unicall 0.0.2 and asterisk 1.0.7
  
   Guillermo
  
  
   From: acriollo [EMAIL PROTECTED]
   Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
   Discussionasterisk-users@lists.digium.com
   To: Asterisk-Users@lists.digium.com
   Subject: [Asterisk-Users] unicall deploy
   Date: Sat, 3 Sep 2005 15:04:20 -0500
   
   Hi every one .
   
   There are any out there that have a unicall deploy working without
 problem
   ?
   Can give me some tips or referenece about his config ?
   
   Regards
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Re: [Asterisk-Users] sending fax

2005-09-05 Thread Harald Klein


Hi Chris, Hi Arne,

Am 5.9.2005 schrieb Chris Shipman [EMAIL PROTECTED]:
I've  seen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.
It would need to be able to print to as local printer and then send out
Asterisk.

What about:

Client with Postscript printer driver
Some kind of a printing system (samba with lpr[ng] and/or cups etc.) to
access the fax-printer via smb/cifs/lpr/ipp/whatever..
Output filter for the fax-printer to convert Postscript to tiff and
generate
a call file with App txfax...

The problem is to tell the printer the number to fax to...
You can grep in the Postscript file for a predefined string (for example
Fax Recpient Nr) and generate some matching templates in your office
suite..

Search for HylaFax solutions, they are pretty much the same...

Hari


Chris

- Original Message -
From: Arne Morten Johansen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 05, 2005 6:27 AM
Subject: SV: [Asterisk-Users] sending fax


 What about faxing yourself if you don't have a scanner?

 -Opprinnelig melding-
 Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Johan van
Tongeren
 Sendt: 5. september 2005 09:11
 Til: Asterisk Users Mailing List - Non-Commercial Discussion
 Emne: RE: [Asterisk-Users] sending fax

 [macro-fax-dialing]
 exten = s,1,SetCIDNum(0${CALLERIDNUM})
 exten = s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t)
 exten = s,3,Goto(900)
 exten = s,103,Goto(900)
 exten = s,900,Busy
 exten = s,901,Hangup

 -Oorspronkelijk bericht-
 Van: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Namens Chris Shipman
 Verzonden: maandag 5 september 2005 7:22
 Aan: Asterisk Users Mailing List - Non-Commercial Discussion
 Onderwerp: [Asterisk-Users] sending fax

 I've read alot on the wiki about sending and receiving faxes thru
 asterisk.
 I've gotten the receive to work great.My question is how does one
 send a
 fax?
 I see lots of instructions about how to send the image to asterisk by
 email,
 etc.  The problem is how does  one make the image of the fax to
 begin
 with?   Has anyone come up with a good solution for this?


 Regards,


 Chris


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Re: [Asterisk-Users] sending fax

2005-09-05 Thread Il Neofita

Hi,
I found on a forum a script that emulate a hylafax this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423
You can use the WHFC in order to send a fax to asterisk.

On 9/5/05, Harald Klein [EMAIL PROTECTED] wrote:
Hi Chris, Hi Arne,Am 5.9.2005 schrieb Chris Shipman [EMAIL PROTECTED]:I'veseen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.It would need to be able to print to as local printer and then send outAsterisk.What about:Client with Postscript printer driver
Some kind of a printing system (samba with lpr[ng] and/or cups etc.) toaccess the fax-printer via smb/cifs/lpr/ipp/whatever..Output filter for the fax-printer to convert Postscript to tiff andgeneratea call file with App txfax...
The problem is to tell the printer the number to fax to...You can grep in the Postscript file for a predefined string (for exampleFax Recpient Nr) and generate some matching templates in your office
suite..Search for HylaFax solutions, they are pretty much the same...HariChris- Original Message -From: Arne Morten Johansen 
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSent: Monday, September 05, 2005 6:27 AM
Subject: SV: [Asterisk-Users] sending fax What about faxing yourself if you don't have a scanner? -Opprinnelig melding- Fra: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] På vegne av Johan vanTongeren Sendt: 5. september 2005 09:11
 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing] exten = s,1,SetCIDNum(0${CALLERIDNUM})
 exten = s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t) exten = s,3,Goto(900) exten = s,103,Goto(900) exten = s,900,Busy exten = s,901,Hangup
 -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] Namens Chris Shipman Verzonden: maandag 5 september 2005 7:22 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] sending fax
 I've read alot on the wiki about sending and receiving faxes thru asterisk. I've gotten the receive to work great.My question is how does one send a fax?
 I see lots of instructions about how to send the image to asterisk by email, etc.The problem is how doesone make the image of the fax to begin with? Has anyone come up with a good solution for this?
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[Asterisk-Users] Asterisk architecture

2005-09-05 Thread housi mueller


I am new with asterisk and hope somebody can help me.

Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.)

Thank you in advace

Housi Mueller

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[Asterisk-Users] Cisco 7960 upgrades

2005-09-05 Thread Sascha Ferley









Hi, 



I got a problem of having to upgrade 35 Cisco 7960 phones
from default firmware of 3.1 to 7.5. 
The problem I get is that when trying to upgrade I see on the tftplog that it
cant seem to find the file (8 character issue). 

So I renamed the files to suit what is supposed to be in
them. 



I am trying incremental upgrades from 3.1 - 5.3 -
7.5, with no luck. It goes to Upgrading Software and sits there endlessly
redownloading the same file. 

It seems to stall going no-where .. Any one successfully
upgraded the phones from default? What are any of the specifics. With the 5x
series do I need the P003-05 in the OS79XX.TXT file or still the P0S3 ? 



Anyone have any ideas as to what I should do? I cant
seem to get the pre 5x versions of the software any more. Seems with my
contract I can only downgrade to 5x series. All that shows on the Cisco CCO
site. 



Please let me know

Thanks



Sascha






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Re: [Asterisk-Users] Asterisk architecture

2005-09-05 Thread Il Neofita

>From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller [EMAIL PROTECTED] wrote:

I am new with asterisk and hope somebody can help me.

Is a configuration like shown on the picture with asterisk
correct? Some phone calls arriving in Branch 1 should be
redirected automatically to Branch 2 and all phone calls made
from Branch 2 should going out over Branch 1. (Branch 2 is not
connected directly with a PSTN.)

Thank you in advace

Housi Mueller

		 Click here to donate to the Hurricane Katrina relief effort.
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Re: [Asterisk-Users] res_features.so (Call Features Resource) not loading

2005-09-05 Thread Michiel van Baak

On 04:48, Tue 06 Sep 05, Asterisk Sales wrote:
 hello everybody,
 i have updated my rpm asterisk to current cvs 1.0.9. I had been using rpm 
 asterisk which comes with suse 9.2. the main reason i updated my asterisk to 
 get the attended transfer feature. i have installed another cvs
 1.0.9asterisk in Redhat 9 and it works perfect.
 here what i found:
  *in suse 9.2:* *CLI show modules
 
 .
 res_features.so Call Parking Resource* Note:* it shows Call Parking Resource
 .
 ..
  *in Redhat 9:*
 *CLI show modules
 .. 
 
 res_features.so Call Features Resource *Note:* it shows Call Features 
 Resource
 ..
 .
  i think asterisk is not loading call features resources in suse 9.2 so i 
 also added a line in
 /etc/asterisk/modules.conf
 
 load =res_features.so 
 .
  still attended tranfer feature is not working. i want to use suse 9.2 as it 
 got IDSN driver support. i use 2 BRI cards. 
 please help
  Thanks in advance
 shaon (AU)

Hi,

The attended transfer stuff is not in 1.0.x
You have to install CVS for that.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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[Asterisk-Users] Assessing network quality

2005-09-05 Thread Chris Mason (Lists)

I am trying to trouble shoot one of my ISP's network and compare to my 
other ISPs offering. Although network 1 is reasonably fast and has low 
enough latency, voice quality is not good and the reason for this is not 
readily apparent using standard network tools.
What tools can be used to assess the quality of the network in terms of 
it's suitability for voice? I am using ping, mtr, smokeping for general 
network reliability and using visualroute to give me info, but I need 
some voice specific quality metrics. Any ideas?


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Zaptel issue

2005-09-05 Thread Asterisk

I'm having a bad time getting 3 TE410P's and one TDM400 working in the same
box.

At some point during the install of second card, wcusb starts loading.  I
believe
this is one of the TE410 Cards causing this as there is no usb enabled.

Module  Size  Used byNot tainted
audit  89880   2  (autoclean)
usbserial  23420   0  (autoclean) (unused)
lp  8964   0  (autoclean)
parport36832   0  (autoclean) [lp]
autofs415832   0  (autoclean) (unused)
tg367368   1
wcusb  19552   0  (unused)
usbcore77376   0  [usbserial wcusb]
wct4xxp70752   0  (unused)
zaptel179872   4  [wcusb wct4xxp]
floppy 56656   0  (autoclean)
sg 36236   0  (autoclean) (unused)
microcode   5688   0  (autoclean)
ext3   85736   2
jbd50668   2  [ext3]
aic7xxx   160880   0  (unused)
diskdumplib 4940   0  [aic7xxx]
sd_mod 13968   0  (unused)
scsi_mod  106664   3  [sg aic7xxx sd_mod]

If I rmmod the usb stuff, it come back on next reboot...
At reboot, zaptel is complaining that modules are busy and can't remove
them.
How can I stop asterisk from loading this ?

Bart



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[Asterisk-Users] Re: Re: Nokia 32 Terminal

2005-09-05 Thread andrutto

AbdelRahman Tarzi ha scritto:

If you wish to connect it to an FXS you will need a special cable which
Nokia sells..
  

you don't really need a special cable for FXS, the cable is a standard 
phone cable with a j11 4/6 pin plug. Just read the tech manual from the 
nokia website for the pinout.

Connecting to an FXO (which expects a line) is the default.
Check the normal stuff (like dialstring) before you suspect the device..
They're really maintenance-free !!
  

I have a problem with the external antenna. No signal gain with it 
connected to the nokia 32 terminal.
You can play with the AT command via serial port to see the signal 
quality level, and I advice you to disable the gsm call waiting service.

Sergio

Hi,

   I tried everything with no success. I even restore the factory defaults but 
without positive effect (call waiting service was disable). The terminal could 
not be broken because I receive calls. I checked the monitor and the signal was 
very strong. Dose anyone have some other tips.

Regards 

Andrutto


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Re: [Asterisk-Users] Cisco 7960 upgrades

2005-09-05 Thread Matthew Boehm

You cannot go from 5.3 - 7.5. You must go from 5.3 - 7.0 then to 7.5.

-Matthew

 From: Sascha Ferley [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Mon, 5 Sep 2005 13:19:40 -0600
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Cisco 7960 upgrades

 Hi, 

 I got a problem of having to upgrade 35 Cisco 7960 phones from default
 firmware of 3.1 to 7.5.
 The problem I get is that when trying to upgrade I see on the tftplog that
 it can't seem to find the file (8 character issue).

 So I renamed the files to suit what is supposed to be in them.

 I am trying incremental upgrades from 3.1 - 5.3 - 7.5, with no luck. It
 goes to Upgrading Software and sits there endlessly redownloading the same
 file. 

 It seems to stall going no-where .. Any one successfully upgraded the phones
 from default? What are any of the specifics. With the 5x series do I need
 the P003-05 in the OS79XX.TXT file or still the P0S3 ?

 Anyone have any ideas as to what I should do? I can't seem to get the pre 5x
 versions of the software any more. Seems with my contract I can only
 downgrade to 5x series. All that shows on the Cisco CCO site.

 Please let me know

 Thanks

 Sascha

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Re: [Asterisk-Users] Cisco 7960 upgrades

2005-09-05 Thread Harald Klein


Am 5.9.2005 schrieb Sascha Ferley [EMAIL PROTECTED]:

Hi,

Hi there,

The problem I get is that when trying to upgrade I see on the tftplog that
it can't seem to find the file (8 character issue).

dunno about your 8 char issue.. use a sane os ;)

It seems to stall going no-where .. Any one successfully upgraded the phones
from default?

yes.

 What are any of the specifics. With the 5x series do I need
 the P003-05 in the OS79XX.TXT file or still the P0S3 ?

[EMAIL PROTECTED]:~$ more /tftpboot/OS79XX.TXT.53
P0S3-05-3-00
[EMAIL PROTECTED]:~$ more /tftpboot/OS79XX.TXT.60
P0S3-06-0-00
[EMAIL PROTECTED]:~$ more /tftpboot/OS79XX.TXT
P003-07-4-00


Anyone have any ideas as to what I should do? I can't seem to get the pre 5x
versions of the software any more. Seems with my contract I can only
downgrade to 5x series. All that shows on the Cisco CCO site.

hmm, i can see all the files on the cco.. oldest is P0S30201.

scroll down ;)
and read the cisco documentation. its fine. should explain all your
firmware/loader issues..

much phun,

Hari


Please let me know

Thanks



Sascha

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Re: [Asterisk-Users] asterisk CAPI dial-in issues