date:20060627


Hi again,

the TR6T parameter (i have german settings for my AMO
so it is TR6Q ;-)) resolved the same issue for my...
the difference is that i have an IP-trunk (using
oh323) between Asterisk and the HiPath. Have you tried
to remove the TR6T parameters...

Can you also paste the following outputs from the H4K

DISPLAY-APS:TYPE=PSGL,SP=y0*;
REGENERATE-TDCSU:PEN1=XX-XX-XX-XX; (change xx-xx-xx-xx
to the pin of the isdn trunk e.g 01-02-25-00) 
DISPLAY-COT:COTNO=XX; (change XX to the cot number of
the trunk)

if the log is not to huge please paste the last 30 min
of the history file. Try to reproduce the issue after
that type:

START-HISTA:RTYPE=SEARCHB,STIME=2006-06-27/09:00,ETIME=2006-06-27/09:30;

adjust the start time and the end time in a way that
the test is in the range between STIME and ETIME...

regards rich...

--- Josué Conti [EMAIL PROTECTED] wrote:

 Hi Richard.
 Thank you very much for its attention. In the
 reality what is occurring is
 that in some originated calls of the HiPath with
 destination to the Asterisk
 they are being without the dumb and rings. I do not
 have this parameter in
 my HiPath 4000, what I have seemed in the COT is
 TR6T (1tr6 isdn tie link)
 would be this parameter?
  Best Regards
 Josué
 
 2006/6/26, richard Coco [EMAIL PROTECTED]:
 
 
  Hi Josué
 
  if the Siemens phone calls Asterisk, it didn't get
 a
  dial tone from Asterisk? Is it correct?
 
  if yes, this is depending of Asterisk which didn't
  generates a ringback messages as it expexts dial
 ton
  generation localy. So try this workaround for
 HiPath
  local dial ton generation:
  - Add option TR6Q(TRGT) to the class of trunk
 (COT)
  parameters
 
  hope it will help...
 
  rich
 
 
 
 
 
  --- Josué Conti [EMAIL PROTECTED] wrote:
 
 Hello all.
I have installed and functioning
 asterisk-1.2.9.1
   where I effected one
   upgrade in asterisk-1.0.9, is interconnected
 with a
   PABX Siemens HiPath 4000
   in ISDN PRI with protocol QSIG, the one that is
   happening he is that the
   calls originated for PABX Siemens and destined
 to
   SIP phones asterisk are
   being without audio, nor Ring, is dumb. They
 could
   help in this case me?
   Best Regards
  
   Josué
   
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[Asterisk-Users] siemens pbx and asterisk

Hello all,I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how?
thanks.Lito
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RE: [Asterisk-Users] Asterisk ACD with Polycom IP501

2006-06-27 Thread Dean @ INKnBITs

I'm new to this and don't know how to do a sip trace, but have attached the
files as requested.

Thanks for your help.
Dean.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke
Sent: 26 June 2006 15:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501


 Hi Dean -

 It should be working. If not, please email me a sip debug trace along
with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf.

 Thanks.

 BJ

On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote:
 Hi,

 Has anybody got the polycom acd function to work? I have the following
 setup:

 Debian 3.1 - 2.6.8 linux
 zlib-1.1.4
 libpri-1.2.3
 zaptel- 1.2.6
 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
 error when doing a make install about needing a newer version of libpri
and
 zaptel, I got the above versions from asterisk.org, are there newer
version
 anywhere else?

 In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo
 (demo as from extensions.conf context)

 I have setup an agent in agents.conf as ,1234,Name

 I have changed in the sip.cfg of the polycom phone:
 feature.15.name=acd-login-logout feature.15.enabled=1
 feature.16.name=acd-agent-availability feature.16.enabled=1

 and in the phone1.cfg of the polycom I'm only using line1 so made the
 changes below:
 reg.1.acd-login-logout=1
 reg.1.acd-agent-available=1


 I get the login button on the phone, and when I try and login with the

 agent it just goes back to login.


 Any help would be appreciated.

 Thanks,
 Dean Bath




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agents.conf
Description: Binary data


sip.conf
Description: Binary data
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Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000


Hi again...

normally the 0/16 is a d-channel.

check the config in the zapata.conf. You should have
some thing like this

/etc/zapata.conf
bchan=1-15 
dchan=16 
bchan=17-31 

/etc/asterisk/zapata.conf
channel = 1-15,17-31 

i don't rember exactelly but in /proc/zaptel there is
the possibility to check if the channels are in use or
not. Maybe someone else can give you a hint...

sorry but i only interconnect Asterisk and H4K using
chan_capi and i have no experience with zapata ;-(

rich

--- Josué Conti [EMAIL PROTECTED] wrote:

 Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386
 pri_dchannel: Ring requested
 on unconfigured channel 0/16 span 1
 I noticed this message in the CLI, when I tried to
 effect one call of HiPath
 4000 for asterisk. Ring occurred, however when the
 voicemail of asterisk
 took care of call it was dumb, without no sound. I
 thank the attention
 Regards
 
 Josué
 
 2006/6/26, Josué Conti [EMAIL PROTECTED]:
 
   Hi Richard.
  Thank you very much for its attention. In the
 reality what is occurring is
  that in some originated calls of the HiPath with
 destination to the Asterisk
  they are being without the dumb and rings. I do
 not have this parameter in
  my HiPath 4000, what I have seemed in the COT is
 TR6T (1tr6 isdn tie link)
  would be this parameter?  
Best Regards
  Josué
 
  2006/6/26, richard Coco [EMAIL PROTECTED]:
 
  
   Hi Josué
  
   if the Siemens phone calls Asterisk, it didn't
 get a
   dial tone from Asterisk? Is it correct?
  
   if yes, this is depending of Asterisk which
 didn't
   generates a ringback messages as it expexts dial
 ton
   generation localy. So try this workaround for
 HiPath
   local dial ton generation:
   - Add option TR6Q(TRGT) to the class of trunk
 (COT)
   parameters
  
   hope it will help...
  
   rich
  
  
  
  
  
   --- Josué Conti [EMAIL PROTECTED] wrote:
  
  Hello all.
 I have installed and functioning
 asterisk-1.2.9.1
where I effected one
upgrade in asterisk-1.0.9, is interconnected
 with a
PABX Siemens HiPath 4000
in ISDN PRI with protocol QSIG, the one that
 is
happening he is that the
calls originated for PABX Siemens and destined
 to
SIP phones asterisk are
being without audio, nor Ring, is dumb. They
 could
help in this case me?
Best Regards
   
Josué

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Re: [Asterisk-Users] x100p buying advice

2006-06-27 Thread Gareth Blades

I would guess the card is actually a http://www.x100p.com/products_1.htm
and may be x100p selling it on ebay themselves.

I have one of these cards itself and it works fine. There is a bit of
echo initially but it gets cancelled our fairly quickly. Apart from
turning on echo cancellation I have not adjusted any other settings.

On Tue, 2006-06-27 at 01:16, Rod Morison wrote:
 I'm looking to get an x100p off ebay and am not particularly familar 
 with the life cycle of the card.. An Authentic X100P listing has a 
 buy it now of $29.95 and says
 
 There are 3 types of cards Asterisk would recognize:
 *Screenshots from the official, original driver install
 Cheap OEM X100P,Clones, Compatibles, Knock-Offs
Found a Wildcard FXO: Generic Clone
 The X101P (note the 101, not 100) is a Low-end version of X100P 
 which uses low grade chips
Found a Wildcard FXO: X101P
 Authentic, Original X100P Speaks for Itself!
Found a Wildcard FXO: X100P
 
 
  From what I gather clone's and knockoffs will have trouble with 
 callerid. Is the Found a Wildcard FXO: X100P enough to establish full 
 featured hardware (assuming an honest seller)? Is there another 
 recommended source besides ebay in this price range?
 
 Thanks
 
 
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[Asterisk-Users] Globe7


Hi all,

Has anybody got an idea if

http://www.globe7.com

supports SIP protocol?

Please send the asterisk config u have it.

Thanks in advance..

Dan
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Re: [Asterisk-Users] DID in United Arab Emirates, Iran, Kuwaiti, Iraq, Bahrain, Jordan, Saudi Arabia.


Hi,

Well to my knowledge, Origination is not legal in these parts of the
world as of now.

Bahrain is open for termination.

http://www.menatelecom.com/products/termination.html

Thanks


Dan


On 27/06/06, Laurent Schweizer [EMAIL PROTECTED] wrote:




Hello,



We are looking for DID in United Arab Emirates, Iran, Kuwaiti, Iraq,
Bahrain, Jordan, Saudi Arabia.



Thanks



 Laurent
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Re: [Asterisk-Users] ASTCC: customer wants 100 accounts

2006-06-27 Thread JP Carballo


Ronald Wiplinger wrote:


I got a request for one customers to set-up 100 accounts.

I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?

To generate 100 cards is not a problem, but if it would work with one 
account number  would be even better


I could use a different context for this customer and use only his 
account code as card number.


Any advice would be appreciated.


I'm not going to ask why the customer needs 100 cards.
If he wants to access them all from 1 account, wouldn't he be happier 
with a single card that has the credits of 100 cards?

In short, an account, not a card. Get my drift?

Or, try making another brand with a markup of 100% I guess. Never tried 
that one though.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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[Asterisk-Users] Avaya 4610sw SIP setup problem

Title: Avaya 4610sw SIP setup problem






Hi all,


I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem!

I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying Registering for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty.

This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address,  Nada.

It is all the more frustrating that everybody seems to have it working easily! Please help.


Here is the contents of my 46xxsettings.txt file :


SET DOMAIN mycompany.com

SET DNSSRVR 204.140.111.43

SET PHNCC 352

SET PHNDPLENGTH 4

SET PHNIC 00

SET PHNOL 0

SET SYSLANG English

SET APPSTAT 1

SET RESTORESTAT 1

SET AGCHAND 0

SET AGCHEAD 0

SET AGCSPKR 0

SET SNTPSRVR 204.140.111.200

SET DSTOFFSET 1

SET DSTSTART 1SunApr2L

SET DSTSTOP LSunOct2L

SET GMTOFFSET -5:00

SET DATESEPARATOR /

SET DATETIMEFORMAT 3

SET DIALPLAN [234]xxx|55

SET DIALWAIT 3

SET MUSICSRVR 

SET MWISRVR 

SET PHNNUMOFSA 3

SET REGISTERWAIT 120

SET SIPDOMAIN sip.mycompany.com

SET SIPPROXYSRVR 204.140.111.219

SET SIPPORT 5070(this is not a typo)

SET SIPREGISTRAR 204.140.111.219

SET SP_DIRSRVR 10.1.1.1

SET SP_DIRSRVRPORT 389

SET SP_DIRTOPDN ou=People,o=avaya.com

IF $MODEL4 SEQ 4602 goto SETTINGS4602

IF $MODEL4 SEQ 4610 goto SETTINGS4610

IF $MODEL4 SEQ 4620 goto SETTINGS4620

IF $MODEL4 SEQ 4621 goto SETTINGS4621

IF $MODEL4 SEQ 4622 goto SETTINGS4622

IF $MODEL4 SEQ 4625 goto SETTINGS4625

IF $MODEL4 SEQ 4630 goto SETTINGS4630

goto END

goto END

SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml

SET WMLPROXY 204.140.111.249

SET WMLPORT 3128

goto END

goto END

goto END

goto END

goto END

SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm

SET PHNEMERGNUM 112

goto END



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Re: [Asterisk-Users] TE420P/TE415P?

2006-06-27 Thread Rob Lith

On 26/06/06, Boris Bakchiev [EMAIL PROTECTED] wrote:
Can the TE406P card's VPM module be swapped for the new revision withOctasic chipset?The VPM450M requires a firmware upgrade to the existing base TE2/4XXP cards. This new firmware is known as 3rd Generation firmware. Digium have an upgrade program and you should be able to organise that with the vendor you got your cards from.
RegardsRob
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[Asterisk-Users] Re: siemens pbx and asterisk

Title: Re: siemens pbx and asterisk






Hi Lito,


We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers.


On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us).

On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk.

I hope this helps,


Silviu



---

Hello all,

I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how?

thanks.

Lito 



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Re: [Asterisk-Users] TE420P/TE415P?

2006-06-27 Thread Rob Lith

On 25/06/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- C F [EMAIL PROTECTED] wrote:Neither. It's a separate device, entirely unrelated to any TDM cards (which means it can be used for any type of channel, not just TDM).
The final specs for the number of channels are not yet determined, but we expect to do at least 100 channels of G.729 and/or G.723.1 per board.Kevin, does the card include the licence for the codecs? Otherwise the card at +- $1994 SRP + codecs is quite expensive?
RegardsRob
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[Asterisk-Users] Re: Asterisk x Siemens HiPath 4000

Title: Re: Asterisk x Siemens HiPath 4000






Hi,


Could you post your /etc/zaptel.conf and zapata.conf?


Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)?


Silviu




Hello all.

I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me?

Best Regards



Josué



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Re: [Asterisk-Users] siemens pbx and asterisk


Hi,

which Hicom and which version is installed?

Hicom 300 or Hicom100?

rich

--- Lito Lampitoc [EMAIL PROTECTED] wrote:

 Hello all,
 
 I'm new to asterisk. Our company wants to setup an
 asterisk server and will
 eventually move to IP centric phones, but they don't
 want to just throw away
 the old Siemens PBX, so during the process we want
 to integrate it with
 asterisk. Is it possible? and how?
 
 thanks.
 
 Lito
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[Asterisk-Users] dss1 progressing message on zap channel

2006-06-27 Thread Rosario Pingaro




I am going to try to figure out why mu asterisk box 
connected by back to back cable to an PRI appliance is not going to send the 
PROGRESSING dss1 message.
In fact i see the SETUP and the follwing CALL 
PROCEEDING but not the PROGRESSING so the appliance doesn't allow the "early 
audio" !

this is my zapata.conf
[channels]context = serswitchtype = 
euroisdnusecallerid = yesechocancel = yesechocancelwhenbridged = 
yesrelaxdtmf=yesrxgain = -1.0txgain = 0.0

signalling = pri_cpegroup =1channel = 
1-15,17-31

;Note: Customers in the UK connected to BT might 
require this 
parameter:pridialplan=localpriindication=inbandcallprogress=yesprogzone=uk

Any comment about it

Reagrds
Rosario

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[Asterisk-Users] Background + Dial

Hi everybody,

I try this :

[incoming_from_fxo_card]
exten = s,1,Answer()
exten = s,2,Background(filename) 
exten = s,3,Dial($(INTERNAL_SIP_TEL))

But * wait the file is finish before make Dial to SIP channel.


Background(filename)  (from voip-info.org)
= Starts playing a given sound file, but immediately
returns, permitting the sound file to play in the background
while the next commands (if any) execute. 

I want to Dial a SIP channel while playing sound and waiting
for a digit from a ZAP channel. In other words, i want to
make a interactive MoH while waiting for the SIP channel answer.
Is it possible?

Thanks a lot and excuse me for my poor english (I'll fix
this in few months).

GL

- ALICE SECURITE ENFANTS -
Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants, le 
contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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[Asterisk-Users] Help Asterisk crashes

2006-06-27 Thread Fredrik Emil Jensen

I am getting thousand of these messages in asterisk console

Jun 27 12:35:55 WARNING[16496]: codec_gsm.c:194 gsmtolin_framein:
Invalid GSM data

And after some time the system crashes. Does anyone know why?

I running Asterisk SVN-trunk-r7522 built

Does it help to upgrade the system?

Regards,
Fredrik Jensen


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[Asterisk-Users] (no subject)

2006-06-27 Thread Vincent renaville

Hi,



I have the same problem with the queue configuration





When I receive 2 calls only 1 phone ring even if more agent's phone are free.



The second call will go to an other agent only if the first call is pickup.



Somebody have a solution ?This is my config file :Queue.conf[general]

;

; Global settings for call queues

;

; Persistent Members

;  Store each dynamic agent in each queue in the astdb so that

;  when asterisk is restarted, each agent will be automatically

;  readded into their recorded queues. Default is 'yes'.

;

persistentmembers = yes

;

; Note that a timeout to fail out of a queue may be passed as part of

; an application call from extensions.conf:

; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])

; example: Queue(dave|t|||45)

autofill = yes



[ticketix]

;

; A sample call queue

;

; Musiconhold sets which music applies for this particular

; call queue (configure classes in musiconhold.conf)

;

autofill=yes

musiconhold = default

;

; An announcement may be specified which is played for the member as

; soon as they answer a call, typically to indicate to them which queue

; this call should be answered as, so that agents or members who are

; listening to more than one queue can differentiated how they should

; engage the customer

;

;announce = queue-ticketix

;

; A strategy may be specified. Valid strategies include:

;

; ringall - ring all available channels until one answers (default)

; roundrobin - take turns ringing each available interface

; leastrecent - ring interface which was least recently called by this queue

; fewestcalls - ring the one with fewest completed calls from this queue

; random - ring random interface

; rrmemory - round robin with memory, remember where we left off last ring pass

;

strategy = roundrobin

;

; Second settings for service level (default 0)

; Used for service level statistics (calls answered within service level time

; frame)

servicelevel = 60

;

; A context may be specified, in which if the user types a SINGLE

; digit extension while they are in the queue, they will be taken out

; of the queue and sent to that extension in this context.

;

;context = qoutcon

;

; How long do we let the phone ring before we consider this a timeout...

;

timeout = 15

;

; How long do we wait before trying all the members again?

;

retry = 5

;

; Weight of queue - when compared to other queues, higher weights get

; first shot at available channels when the same channel is included in

; more than one queue.

;

;weight=0

;

; After a successful call, how long to wait before sending a potentially

; free member another call (default is 0, or no delay)

;

wrapuptime=15

;

; Maximum number of people waiting in the queue (0 for unlimited)

;

maxlen = 0

;

;

; How often to announce queue position and/or estimated holdtime to caller (0=off)

;

announce-frequency = 90

;

;

; How often to make any periodic announcement (see periodic-announce)

;

periodic-announce-frequency=60

;

; Should we include estimated hold time in position announcements?

; Either yes, no, or only once.

; Hold time will be announced as the estimated time,

; or less than 2 minutes when appropriate.

;

announce-holdtime = yes



;

; What's the rounding time for the seconds?

; If this is non-zero, then we announce the seconds as well as the minutes

; rounded to this value.

;

announce-round-seconds = 10

;

; Use these sound files in making position/holdtime announcements. The

; defaults are as listed below -- change only if you need to.

;

queue-youarenext = queue-youarenext  ;   (You are now first in line.)

queue-thereare = queue-thereare;   (There are)

queue-callswaiting = queue-callswaiting ;   (calls waiting.)

queue-holdtime = queue-holdtime;   (The current est. holdtime is)

queue-minutes = queue-minutes ;   (minutes.)

queue-seconds = queue-seconds ;   (seconds.)

queue-thankyou = queue-thankyou;   (Thank you for your patience.)

queue-lessthan = queue-less-than;   (less than)

queue-reporthold = queue-reporthold  ;   (Hold time)

periodic-announce = queue-periodic-announce  ;   (All reps busy / wait for next)

;

; Calls may be recorded using Asterisk's monitor resource

; This can be enabled from within the Queue application, starting recording

; when the call is actually picked up; thus, only successful calls are

; recorded, and you are not recording while people are listening to MOH.

; To enable monitoring, simply specify monitor-format; it will be disabled

; otherwise.

;

; You can specify the monitor filename with by calling

;  Set(MONITOR_FILENAME=foo)

; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}

;

monitor-format = wav49

;

; If you wish to have the two files joined together when the call ends, set this

; to yes.

;

monitor-join = yes

;

; This setting controls whether callers can join a queue with no members. There

; are three

Re: [Asterisk-Users] Zaptel answering the Line

2006-06-27 Thread Thomas Kenyon

Tzafrir Cohen wrote:
 On Sun, Jun 25, 2006 at 08:28:35PM +0100, Thomas Kenyon wrote:
   
 I have a TDM400 card with 3x FXO and 1x FXS ports on it.

 At the moment I'd prefer (till I can get it working more reliable with
 iaxmodem), for a faxmodem to answer one of the lines instead of the
 linecard.

 I've tried changing the context of that line so that the exten = s does
 nothing, but that stops the line from being able to receive calls (get a
 recorded This number is not accepting calls at the moment).

 So my 2 questions are...

 How do I set one of the channels in zaptel.conf (or elsewhere) so that
 it is only available for making calls (and not receiving them).
 

 What do you want to happen to incoming calls? send that specific channel
 (using context= ) to a context that does not even Answer() the line.

 in /etc/asterisk/zapata.conf:

 signalling=fxs_ks
 context=noanswer
 channel = 3
 ; make sure you write another 'context=' after than before the next
 ; channel

 in extensions.conf: I'm not totally sure it is necessary to make this
 explicit, but this will reduce the warnings you'll get in the console:

 [noanswer]
 exten = s,1,Hangup()
   
I tried that and it made the line unobtainable, iaxmodem seems to be
receiving faxes reliably on the other 2 lines, when I'm in over the
weekend I will try and tweak the settings for the remaining line to see
if I can get it reliably working on that line (would be nice to only
have the line card and ADSL modem on that line).

Thanks for your help.


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Re: [Asterisk-Users] best hardphone for Asterisk?

2006-06-27 Thread Matthias Fechner

Hi,

Cullin J. Wible wrote:
 We have also deployed a dozen of the Linksys SPA-1001 single-line FXS
 adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy
 to deploy - $60-$70 US each.

I bought a Grandstream GXP-2000 and played now a little with it. It
seems to work really perfect. The Quality is compareable to my tiptel195
ISDN phone.
The configuration can be done via TFTP and Web.
I ordered it in Germany and the price was 97 EUR inclusive shipping.

The only disadvantage the phone has is the very basic addressbook, but I
think it will be improved with the next firmware versions.


Best regards,
Matthias

-- 

Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots. So far, the universe is winning. --
Rich Cook

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RE: [Asterisk-Users] Background + Dial

2006-06-27 Thread Hoa Thai Duy

Hi GL

Pls. config MOH and use Dial command with m option.
This will allow you execute Dial command while providing Music in the
background.

Hope it help

Hoa Thai Duy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 5:01 PM
To: asterisk-users
Subject: [Asterisk-Users] Background + Dial

Hi everybody,

I try this :

[incoming_from_fxo_card]
exten = s,1,Answer()
exten = s,2,Background(filename)
exten = s,3,Dial($(INTERNAL_SIP_TEL))

But * wait the file is finish before make Dial to SIP channel.


Background(filename)  (from voip-info.org) = Starts playing a given sound
file, but immediately returns, permitting the sound file to play in the
background while the next commands (if any) execute. 

I want to Dial a SIP channel while playing sound and waiting for a digit
from a ZAP channel. In other words, i want to make a interactive MoH while
waiting for the SIP channel answer.
Is it possible?

Thanks a lot and excuse me for my poor english (I'll fix this in few
months).

GL

- ALICE SECURITE ENFANTS - Protégez
vos enfants des dangers d'Internet en installant Sécurité Enfants, le
contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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Re: [Asterisk-Users] Meetme max users

2006-06-27 Thread Kai Ober


Today i put 10 users in a Meetme on a 700MHz machine.
but the result did not satisfy me.

I had all 10 Phones in front of me, cause i'm testing my asterisk.
so i could speak on one phone and listen on any other.
i had a delay of  1 sec of my spoken word(s)

so i think, that you should use a BIG CPU.

a friend of mine mentioned, that if there are many SIP participiants in 
the conference the delay will even be greater.

(one guy told a joke, and laughing was delayed from 10 to 30 seconds)

regards
Kai

thanks

we are planing to have around 50-60 users in 1 room.
  


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RE: [Asterisk-Users] Background + Dial

Thanks for your reply,

But I want to have an interactive menu, not just a music.
So, the customer can have information menu while he's
waiting the call is answer. 
I'dont now if it's possible with MoH.

Thanks a lot

-- Initial Header ---

From  : [EMAIL PROTECTED]
To  : 'Asterisk Users Mailing List - Non-Commercial
Discussion'asterisk-users@lists.digium.com
Cc  : 
Date  : Tue, 27 Jun 2006 18:58:26 +0700
Subject : RE: [Asterisk-Users] Background + Dial

Hi GL

Pls. config MOH and use Dial command with m option.
This will allow you execute Dial command while providing
Music in the
background.

Hope it help

Hoa Thai Duy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 5:01 PM
To: asterisk-users
Subject: [Asterisk-Users] Background + Dial

Hi everybody,

I try this :

[incoming_from_fxo_card]
exten = s,1,Answer()
exten = s,2,Background(filename)
exten = s,3,Dial($(INTERNAL_SIP_TEL))

But * wait the file is finish before make Dial to SIP channel.


Background(filename)  (from voip-info.org) = Starts playing
a given sound
file, but immediately returns, permitting the sound file to
play in the
background while the next commands (if any) execute. 

I want to Dial a SIP channel while playing sound and waiting
for a digit
from a ZAP channel. In other words, i want to make a
interactive MoH while
waiting for the SIP channel answer.
Is it possible?

Thanks a lot and excuse me for my poor english (I'll fix
this in few
months).

GL

- ALICE SECURITE ENFANTS
- Protégez
vos enfants des dangers d'Internet en installant Sécurité
Enfants, le
contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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- ALICE SECURITE ENFANTS -
Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants, le 
contrôle parental d'Alice.
http://www.aliceadsl.fr/securitepc/default_copa.asp


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Re: [Asterisk-Users] Meetme max users

2006-06-27 Thread Matt Florell


700MHz is very underpowered for a server that will do a lot of meetme.
I would recommend at least a 1.6GHz P4

As for the delay, that problem is usually made worse by using ztdummy
with poor zttest scores. try a different ztdummy timer source or a
hardware zaptel timer if possible because that's what meetme uses to
keep track of timing and sending out frames.

MATT---

On 6/27/06, Kai Ober [EMAIL PROTECTED] wrote:

Today i put 10 users in a Meetme on a 700MHz machine.
but the result did not satisfy me.

I had all 10 Phones in front of me, cause i'm testing my asterisk.
so i could speak on one phone and listen on any other.
i had a delay of  1 sec of my spoken word(s)

so i think, that you should use a BIG CPU.

a friend of mine mentioned, that if there are many SIP participiants in
the conference the delay will even be greater.
(one guy told a joke, and laughing was delayed from 10 to 30 seconds)

regards
Kai
 thanks

 we are planing to have around 50-60 users in 1 room.


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Re: [Asterisk-Users] siemens pbx and asterisk

2006-06-27 Thread Viktor Tatianin




Hello Lito
My PBX HICOM 350 interconnect with asterisk via 
Tormenta cards use e1withEDSS1 
protocol this work fine

  - Original Message - 
  From: 
  Lito 
  Lampitoc 
  To: asterisk-users@lists.digium.com 
  
  Sent: Tuesday, June 27, 2006 11:18 
  AM
  Subject: [Asterisk-Users] siemens pbx and 
  asterisk
  Hello all,I'm new to asterisk. Our company wants to 
  setup an asterisk server and will eventually move to IP centric phones, but 
  they don't want to just throw away the old Siemens PBX, so during the process 
  we want to integrate it with asterisk. Is it possible? and how? 
  thanks.Lito
  
  

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Re: [Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

2006-06-27 Thread Bruce Reeves

The Wall Street Journal had a write up on this and after reading through it I did not see much in the way of improvement. It seems like the main focus of Micro$oft is to integrate their products with phone systems which can already be done. The article talked about dialing from Office apps and voicemail in email which can all be accomplished in * and maybe in other products like Cisco and Avaya for a price.
On 6/27/06, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote:
On Tue, June 27, 2006 0:26, shadowym said: They have been talking about this for awhile.If you look at the real time and embedded operating system world they have not really done so well over
 the many years they have been trying. Just throwing money at the problem has never worked for them in the past either.Perhaps because people expect devices like that to Just Work(tm),something Embedded Linux is better known for than Embedded Windows is?...
 The Asterisk community has nothing to worry about in the near term if ever IMHO.Unless they buy Digium... That'd give them a serious amount of code toobfuscate and hide in closed source products! ;-)
--F PeetersPIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN2 Sweex HFC-PCI cards___
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Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000

2006-06-27 Thread Josué Conti

Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us
#zapata.conf
[trunkgroups]
[channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable = yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=no
callwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes
rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31

Best Regards

Josué

2006/6/27, Herchi Silviu [EMAIL PROTECTED]:



Hi, 
Could you post your /etc/zaptel.conf and zapata.conf? 
Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)? 
Silviu 
 
Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9
, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me?



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[Asterisk-Users] Callstatus on bridge IAX2 - ZAPTEL is always answer even if the call fails

2006-06-27 Thread Mouss Greg

Hello,
In my asterisk box, i have a zaptel card connected to my analogic pstn line.
I'm using a IAX2 client to call outside :

IAX2 client -- Asterisk -- Zaptel card  France telecom line

When checking cdr logs file, i always have an ANSWER on call
status when call on this trunk, even if the final destination
does not answer. Is ANSWER the status of the bridge between channels
IAX2  ZAP ? or is it really the final status of the link ?

I'm looking forward to get the final status of the call.
If somebody could help.
Thanks.
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[Asterisk-Users] WebPhone

2006-06-27 Thread Il Neofita

Hi,someone know a good webphone, possibily a free oneThx
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Re: [Asterisk-Users] Re: siemens pbx and asterisk

2006-06-27 Thread Michael Hamann

Hi Silviu,

did you manage to get the callername to work? I have a comparable setup
with a hipath System but I can´t get the callername to be displayed over
the trunk. The callernumber works but not the name...

Any suggestion?

Thanks
Michael


 We have successfully integrated an existing Siemens HiPath 4500 PBX with
 two Asterisk servers.

 On the first one we use a H.323 trunk (it needs a card on the PBX, I
 think it's called HG3550). It works pretty well, except for one missing
 feature - the callerid name is not transmitted over the link (it is a
 limitation of the PBX that should disappear when it is upgraded to the
 V3 version). The nice thing is it doesn't take any special hardware on
 the Asterisk server - you just have to compile and setup an H.323
 channel (asterisk-oh323 works best for us).

 On the second one we have a Digium TE110P connected to the PBX using a
 PRI. It works well too, you just need the PBX to have a trunk defined
 and you're ready to go. We only use ten channels, so I can't say if the
 performance is better. In this case you need libpri and zaptel on the
 Asterisk.

 I hope this helps,

 Silviu


 ---
 Hello all,

 I'm new to asterisk. Our company wants to setup an asterisk server and
 will eventually move to IP centric phones, but they don't want to just
 throw away the old Siemens PBX, so during the process we want to
 integrate it with asterisk. Is it possible? and how?
 thanks.
 Lito


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[Asterisk-Users] Problem with callerid in sip to isdn gateway

2006-06-27 Thread Morten Isaksen

Hi!

I have this setup:

PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO
Digium TE410P is used in both Asterisk 1 and 2.

When I set the CLIR bit on the PABX the Callerid / ANI is removed somewhere between the SIP interface on Asterisk 1 and the SIP interface on Asterisk 2.

I need the callerid / ANI on Asterisk 2 in order for the TELCO to bill me correctly.

Is there any way I can tell Asterisk 1 to keep the callerid and the clir bit, and then let Asterisk 2 deal with it?
-- Morten Isaksenhttp://www.misak.dk/blog/ 
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Re: [Asterisk-Users] Callstatus on bridge IAX2 - ZAPTEL is always answer even if the call fails

2006-06-27 Thread John Novack


Mouss Greg wrote:


Hello,
In my asterisk box, i have a zaptel card connected to my analogic pstn 
line.

I'm using a IAX2 client to call outside :

IAX2 client -- Asterisk -- Zaptel card   France telecom line

When checking cdr logs file, i always have an ANSWER on call status  
when call on this trunk, even if the final destination does not 
answer. Is ANSWER the status of the bridge between channels IAX2  
ZAP ? or is it really the final status  of the link ?


I'm looking forward to get the final status of the call.
If somebody could help.
Thanks


Zap channels consider the call answered once dialing is complete, , 
which complicates billing, and any hope of another action on a real no 
answer.
The writers never made any provisions for lines that provide any sort of 
answer supervision


John Novack

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Re: [Asterisk-Users] Problem with callerid in sip to isdn gateway

2006-06-27 Thread trixter aka Bret McDanel

On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote:
 Hi!
  
 I have this setup:
  
 PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO
 
 Digium TE410P is used in both Asterisk 1 and 2.
  
 When I set the CLIR bit on the PABX the Callerid / ANI is removed
 somewhere between the SIP interface on Asterisk 1 and the SIP
 interface on Asterisk 2.
  
Have you used a packet sniffer to ensure that its actually sent to
asterisk 2?  If it isnt then that may be the entire problem.  Before
trying to diagnose anything on the isdn side I would make sure that it
is infact being sent correctly.  Alternatively you can try some noops()
on asterisk2 for when a call is received to display the caller id to the
console, that may be easier for some than reading sip headers.


 
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058  US WA +1 360 207 0479
US NY +1 516 687 5200  FreeWorldDialup: 635378
http://www.trxtel.com the VoIP provider that pays you!


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[Asterisk-Users] Callstatus on bridge IAX2 - ZAPTEL is always answer even if the call fails

2006-06-27 Thread Mouss Greg

Thanks John for your quick answer,

You're right, i'm trying to put a strong billing system in place, after
many months using ASTCC, i'm now integrating a2billing which seems to
be stronger (my own opinion). But i did'nt clearly undertstand what you
said (my poor english level again ...) Could have few minutes to
clarify ? Thanks again.

Mouss.

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Re: [Asterisk-Users] [ISSUE] Unable to divert external calls.

2006-06-27 Thread Steve Davies


On 6/26/06, Peter J Dean [EMAIL PROTECTED] wrote:

I have a issue trying to understand why Asterisk-PBX, when a SNOM
(320 or 360) successfully redirects/diverts a call when it is a local
extension, but fails when you enter external number.
Both the local extension dial and external extension dial are within
the same context [from-sip] and both phones are capable of making
external calls.



There is probably some difference between the 302 Moved SIP messages
being sent back by the snom in each case.

Your logs seem to imply that:
 For the extension, it is asking for [EMAIL PROTECTED]
 For the external number is is asking for SIP/[EMAIL PROTECTED]

I am not sure why this is, but you /are/ running a pre-release
firmware version, so it may be a known issue?

Steve
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RE: [Asterisk-Users] Re: siemens pbx and asterisk

Hi,

As I wrote, the HiPath needs to be upgraded to version 3 (don't ask me any 
details, I'm not a Siemens expert) in order to have the CallerID name passed 
over the H.323 link. Earlier versions (my case) ony sends and accepts the 
CallerId number.

I have set up a workaround for calls coming to Asterisk: an AGI script sets the 
CallerID name according to their CallerID number by looking it up in a 
database. This is done in real time for every incoming call. Obviously it 
doesn't work for calls going from Asterisk to the HiPath.

Regards,

Silviu

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Hamann
Sent: 27 June 2006 14:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: siemens pbx and asterisk

Hi Silviu,

did you manage to get the callername to work? I have a comparable setup with a 
hipath System but I can´t get the callername to be displayed over the trunk. 
The callernumber works but not the name...

Any suggestion?

Thanks
Michael


 We have successfully integrated an existing Siemens HiPath 4500 PBX 
 with two Asterisk servers.

 On the first one we use a H.323 trunk (it needs a card on the PBX, I 
 think it's called HG3550). It works pretty well, except for one 
 missing feature - the callerid name is not transmitted over the link 
 (it is a limitation of the PBX that should disappear when it is 
 upgraded to the
 V3 version). The nice thing is it doesn't take any special hardware on 
 the Asterisk server - you just have to compile and setup an H.323 
 channel (asterisk-oh323 works best for us).

 On the second one we have a Digium TE110P connected to the PBX using a 
 PRI. It works well too, you just need the PBX to have a trunk defined 
 and you're ready to go. We only use ten channels, so I can't say if 
 the performance is better. In this case you need libpri and zaptel on 
 the Asterisk.

 I hope this helps,

 Silviu


 ---
 Hello all,

 I'm new to asterisk. Our company wants to setup an asterisk server and 
 will eventually move to IP centric phones, but they don't want to just 
 throw away the old Siemens PBX, so during the process we want to 
 integrate it with asterisk. Is it possible? and how?
 thanks.
 Lito


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RE: [Asterisk-Users] SE Michigan asterisk users group

2006-06-27 Thread jglucky

I am in the Cadillac area and would be interested in joining this group
depending on where in the SE it is located and the time the meetings occur.

Thank you,

Jyran Glucky
Advisory Programmer
BlueWare, Inc.
Strategic HealthWare Solutions
3060 W. 13th Street
Cadillac, MI 49601
Phone:  (231) 779-0224 ext. 111
Fax: 231-779-1002
Skype: Jyran Glucky
AIM: JyranGlucky
mailto:[EMAIL PROTECTED]
http://www.blueware.net

BlueWare Fast Facts...
Interoperability Leader Recognized at HIMSS 2006
IBM Beacon Award Winner - Best DB2 Document  Management Solution Worldwide
Awarded 2006 Top 50 Growing Companies in Michigan
A Decade of Delivering the First Best-of-Breed EHR Solution to the Market
IBM Premier Business Partner


   
 Carlos Alperin  
 [EMAIL PROTECTED] 
 om.netTo 
 Sent by:  'Asterisk Users Mailing List - 
 asterisk-users-bo Non-Commercial Discussion' 
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 06/26/2006 11:58  RE: [Asterisk-Users] SE Michigan
 AMasterisk users group
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Ok, I count at least 4. Just lets propose when  where for the first
meeting group, and start to think about issues discussion.

Tom, what to we need for the mailing list? I can do something about that.

Carlos Alperin



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Radon
Sent: Monday, June 26, 2006 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SE Michigan asterisk users group

I'm also in the area, near Southfield.  I'd be interested as well.

On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote:
  I am thinking of getting an asterisk user group together for either SE
  Michigan or just Metro-Detroit.

  How much interest in asterisk in Michigan is there on this list?

  I am already on the board of glimasoutheast, with is a group for
  technology professionals. (very broad range)
  It is a spin-off from Automation Alley, which is SE Michigan's version
  of Silicone Valley.

  --
  Steven

  http://www.glimasoutheast.org

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Is it something someone said, was it something someone said?
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[Asterisk-Users] Re: SE Michigan asterisk users group

2006-06-27 Thread Steven

We should figure out what we are doing before we do any investing in domain
names, etc.

I believe that we have the option to be a sub-group off of the glima network if
desired.
http://www.glima.org/autoalley/GLIMA+Network/Member+Benefits/
They are a non-profit org for technology people in Michigan.
They are backed by Automation Alley and have resources if needed. (meeting
rooms, non-profit paperwork help, etc.)
It may also help with membership, as we would be able to post our meetings on
their calendar.
http://www.glima.org/autoalley/GLIMA+Network/Calendar+of+Events/Calendar.htm?ViewMonth=6-27-2006SubBrandId=0

Carlos, If you would like to set up a temp. mailinglist, feel free. (this will
keep us from clogging this list.)

Please let me know if you guys think I should pursue starting the group as a
subgroup of glima.

--
--
Steven

http://www.glimasoutheast.org

Carlos Alperin [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
OK,

This is what I get till now:

Ron Kushner Sterling Heigths [EMAIL PROTECTED]
Rusty Dekema Ann Arbor [EMAIL PROTECTED]
Jon Radon Southfield [EMAIL PROTECTED]
Steven BerkHolz [EMAIL PROTECTED]
Tom Hayden Livonia [EMAIL PROTECTED]
Michael George Lansing [EMAIL PROTECTED]
Tim Sharp Livonia [EMAIL PROTECTED]
Bradley Watkins ?? [EMAIL PROTECTED]
albeit ?? [EMAIL PROTECTED]
Carlos Alperin Southfield [EMAIL PROTECTED]

If you guys want to start planning on how to start, then is time to start
exchanging direct mails with more info
In order to arrange future steps.

I can create the mailing list, for that I can use my own domain
calperin.com, or we going to need to register a domain on Network Solutions

Waiting for suggestions,

Regards,

Carlos Alperin

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Re: [Asterisk-Users] Meetme max users

2006-06-27 Thread Patrick

On Tue, 2006-06-27 at 14:03 +0200, Kai Ober wrote:
 Today i put 10 users in a Meetme on a 700MHz machine.
 but the result did not satisfy me.
 
 I had all 10 Phones in front of me, cause i'm testing my asterisk.
 so i could speak on one phone and listen on any other.
 i had a delay of  1 sec of my spoken word(s)
 
 so i think, that you should use a BIG CPU.
 
 a friend of mine mentioned, that if there are many SIP participiants in 
 the conference the delay will even be greater.
 (one guy told a joke, and laughing was delayed from 10 to 30 seconds)

Afaik MeetMe delay issues were solved in 1.2.9.1 or a more recent
version of the 1.2 branch. In addition to Matt's remarks about the need
for a good timing source, if you use zaptel, make sure it has the PLL
patch from #6631.

Regards,
Patrick


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Re: [Asterisk-Users] Re: siemens pbx and asterisk

Hello Silviu,Thank you very much for your reply. I will try that.On 6/27/06, Herchi Silviu [EMAIL PROTECTED]
wrote:

Hi Lito,

We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers.

On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an
H.323 channel (asterisk-oh323 works best for us).

On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk.

I hope this helps,

Silviu

---

Hello all,

I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how?

thanks.

Lito

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[Asterisk-Users] voicemail number of recorded messages

2006-06-27 Thread Khaled Chehab









How can I limit extension voicemail messages to 10 messages
per user ?














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[Asterisk-Users] voicemail number of recorded messages

2006-06-27 Thread Khaled Chehab











How can I limit extension voicemail messages to 10
messages per user ?














*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.
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[Asterisk-Users] asterisk to mobile phone

Is it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup?
Thanks in advance.Lito
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RE: [Asterisk-Users] asterisk to mobile phone

2006-06-27 Thread Roger Workman









I think this could be implemented via
follow-me feature of Asterisk







Roger Workman

Business Development

Upperclassman/Universal Holdings LLC

Voice:304.324.3800

Fax:304.324.3801 

ICQ: 4447584

Website: http://www.upperclassman.net

Billing Questions: billing at upperclassman.net

Rental Questions: rentals at upperclassman.net

Maintenance: help at upperclassman.net





This e-mail and any of its attachments may contain sensitive
information, which is privileged, confidential, or subject to copyright
belonging to RW Management Inc, Universal Holdings LLC or Upperclassman LLC.
This e-mail is intended solely for the use of the individual or entity to which
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please notify the sender immediately and permanently delete the original and
any copy of or printout of this e-mail.













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Lito Lampitoc
Sent: Tuesday, June 27, 2006 9:59
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk
to mobile phone





Is it possible to trunk hunt mobile phones in asterisk? say I have one
trunkline and 10 mobile phones brought by the engineers in the field, when
someone calls the trunkline, asterisk will hunt which of the 10 mobile phones
is available. What do I need for this setup? 

Thanks in advance.

Lito





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[Asterisk-Users] Call length limitation

HiI have a problem with Dial application. The dialplan looks like this:;exten = x,1,Dial(Sip/|30|L(6:3:1))exten = x,2,Hangup()exten = h,1,DadAGI()
;The call is limited to 60 sec and after that time the conversation stops, but Asterisk never reach the h extension.I could use the S() option in Dial application but I want to have the announcements. Is there any way to force Asterisk to execute the DeadAGI. Or maybe there is some other solution to achieve a goal.
Any help would be appreciated.CheersAndrew
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RE: [Asterisk-Users] asterisk to mobile phone

2006-06-27 Thread Colin Anderson




A GSM 
gateway will allow you to specify a ruleset so a channel on the gateway is 
always locked to a particular mobile number, then you just send the call from 
Asterisk to the gateway and it will do the hunt for you. 

  -Original Message-From: Lito Lampitoc 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, June 27, 2006 7:59 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] asterisk to mobile phoneIs it possible 
  to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 
  mobile phones brought by the engineers in the field, when someone calls the 
  trunkline, asterisk will hunt which of the 10 mobile phones is available. What 
  do I need for this setup? Thanks in 
advance.Lito
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RE: [Asterisk-Users] Re: siemens pbx and asterisk


hi all,

The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
I'am not sure but i thing that the feature CallerID
Name was introduced in version 3 of the H.323
standard. More informations about the owerviews at
http://www.packetizer.com/voip/h323/.

-Concerning HiPathv3.0.
In version 3.0 the HiPath has a new board (the HG3540)
which supports SIP (for Endpoints) and SIPQ for
SIP-trunking. You are now able to interconnect
Asterisk and HiPath using H.323, ISDN and/or SIPQ.

rich

--- Herchi Silviu [EMAIL PROTECTED] wrote:

 Hi,
 
 As I wrote, the HiPath needs to be upgraded to
 version 3 (don't ask me any details, I'm not a
 Siemens expert) in order to have the CallerID name
 passed over the H.323 link. Earlier versions (my
 case) ony sends and accepts the CallerId number.
 
 I have set up a workaround for calls coming to
 Asterisk: an AGI script sets the CallerID name
 according to their CallerID number by looking it up
 in a database. This is done in real time for every
 incoming call. Obviously it doesn't work for calls
 going from Asterisk to the HiPath.
 
 Regards,
 
 Silviu
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Michael Hamann
 Sent: 27 June 2006 14:58
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Cc: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Re: siemens pbx and
 asterisk
 
 Hi Silviu,
 
 did you manage to get the callername to work? I have
 a comparable setup with a hipath System but I
canï¿½t
 get the callername to be displayed over the trunk.
 The callernumber works but not the name...
 
 Any suggestion?
 
 Thanks
 Michael
 
 
  We have successfully integrated an existing
 Siemens HiPath 4500 PBX 
  with two Asterisk servers.
 
  On the first one we use a H.323 trunk (it needs a
 card on the PBX, I 
  think it's called HG3550). It works pretty well,
 except for one 
  missing feature - the callerid name is not
 transmitted over the link 
  (it is a limitation of the PBX that should
 disappear when it is 
  upgraded to the
  V3 version). The nice thing is it doesn't take any
 special hardware on 
  the Asterisk server - you just have to compile and
 setup an H.323 
  channel (asterisk-oh323 works best for us).
 
  On the second one we have a Digium TE110P
 connected to the PBX using a 
  PRI. It works well too, you just need the PBX to
 have a trunk defined 
  and you're ready to go. We only use ten channels,
 so I can't say if 
  the performance is better. In this case you need
 libpri and zaptel on 
  the Asterisk.
 
  I hope this helps,
 
  Silviu
 
 
  ---
  Hello all,
 
  I'm new to asterisk. Our company wants to setup an
 asterisk server and 
  will eventually move to IP centric phones, but
 they don't want to just 
  throw away the old Siemens PBX, so during the
 process we want to 
  integrate it with asterisk. Is it possible? and
 how?
  thanks.
  Lito
 
 
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Re: [Asterisk-Users] Call length limitation

2006-06-27 Thread El Flynn


Andrew Nowrot wrote:

Hi

I have a problem with Dial application. The dialplan looks like this:

;

exten = x,1,Dial(Sip/|30|L(6:3:1))
exten = x,2,Hangup()

exten = h,1,DadAGI()

;

The call is limited to 60 sec and after that time the conversation stops,
but Asterisk never reach the  h extension.
I could use the S() option in Dial application but I want to have the
announcements. Is there any way to force Asterisk to execute the 
DeadAGI. Or

maybe there is some other solution to achieve a goal.



What does the CLI show when you make the call? That might help in diagnosing 
your problem.


Flynn


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Re: [Asterisk-Users] voicemail number of recorded messages

2006-06-27 Thread El Flynn


Khaled Chehab wrote:
 
How can I limit extension voicemail messages to 10  messages per user ?




If you look in the voicemail.conf.sample file in the source, you can find the 
following lines:



; Maximum number of messages per folder.  If not specified, a default value
; (100) is used.  Maximum value for this option is .
;maxmsg=100

So you would set this to 10 for your voicemail.conf

Cheers,
Flynn


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Re: [Asterisk-Users] Callstatus on bridge IAX2 - ZAPTEL is always answer even if the call fails

2006-06-27 Thread John Novack


Put simply:
Send a call to the PSTN ( analog)  through a TDM 400, Sangoma A200, or 
an X100 card, once the dialing string is sent, the call will report as 
answered.

That's it. You will have no way to PROPERLY bill for the call.
You can assume a call duration of less than one minute ( or whatever 
number you like ) has gone unanswered, but no way to know for sure.


John Novack


Mouss Greg wrote:


Thanks John for your quick answer,

You're right, i'm trying to put a strong billing system in place, 
after many months using ASTCC, i'm now integrating a2billing which 
seems to be stronger (my own opinion). But i did'nt clearly 
undertstand what you said (my poor english level  again ...) Could 
have few minutes to clarify ? Thanks again.


Mouss.



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[Asterisk-Users] isdn-data over iax

2006-06-27 Thread DRi

is it possible to route an ISDN-Data channel over an iax-connection ?

the setup is 

pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk 
Server2 (E1)-connecting to an external isdn-dialin router

via the iax-line the call is transfered as speech which is not accepted at 
the remote end

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RE: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000




Hello,

The main differences I can see:

- in zaptel.conf
you have span=1,0,0,ccs,hdb3, which means you ask Asterisk 
to serve as a timer for the PBX - on my setup the PBX is the master clock and 
Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use 
CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4)

- in zapata.conf
I have switchtype=EuroISDN. Generally speaking, try using 
other switchtypes.

Regards,

Silviu



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Josué 
ContiSent: 27 June 2006 14:41To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: 
Asterisk x Siemens HiPath 4000

Silviu, 
thank's will be this attention. Below my configurations of zapata.conf and 
zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us
#zapata.conf
[trunkgroups]
[channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable 
= 
yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes 
rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31

Best Regards

Josué

2006/6/27, Herchi Silviu [EMAIL PROTECTED]: 


  
  
  Hi, 
  Could you post your /etc/zaptel.conf and 
  zapata.conf? 
  Also, is everything OK the other way round (i.e., 
  from the SIP phones to the PBX)? 
  Silviu 
   
  Hello 
  all. I have installed and functioning 
  asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is 
  interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, 
  the one that is happening he is that the calls originated for PABX Siemens and 
  destined to SIP phones asterisk are being without audio, nor Ring, is dumb. 
  They could help in this case me? 
  
  
  Best Regards  Josué 
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[Asterisk-Users] can Asterisk act as a H.323 Gatekeeper.

2006-06-27 Thread Pawel

Hallo.
I managed to configure asterisk to act as H.323 gateway using asterisk built in 
support for H.323. I found it in ./channels/h323 directory of asterisk sources.

I wonder whether asterisk can play a role of H.323 gatekeeper. If Yes, could 
You tell me some hints on how to do that.

Greetings.
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[Asterisk-Users] F3000 registering to asterisk

2006-06-27 Thread Matt


Hi,
I have an F3000 phone that I am trying to register to asterisk.   As
far as I can tell I have everything in correct.   Are there any little
quirks I need to worry about?  The phone has internet access, set it's
time.. I can access the web config, but it just won't register with
asterisk.   I don't see anything meaningful in the full log.
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[Asterisk-Users] 7960 help: transferring calls

2006-06-27 Thread Chris Bagnall

Greetings all,

Not specifically an asterisk query, but a couple of transfer queries that
I'm sure are obvious to folks who use these phones all the time:

1) how does one do a blind transfer? When a call is answered and one hits
the transfer button, followed by an extension, one has to wait for the other
party to answer, then hit transfer again, before the call is released. I'm
sure there must be an option to answer a call, then fire it straight off to
another extension without waiting for an answer?

2) if there are 2 incoming calls currently on the go (i.e. the first one has
been put on hold for the operator to answer the second call), how does one
determine which call will be transferred when the transfer button is
pressed?  Is there a way to select the source call for a transfer prior to
hitting transfer?

3) when handling 2 calls, how does one swap between them?

These phones are running sccp through chan_sccp if that makes any difference
to operation.

Thanks in advance folks.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] WebPhone

2006-06-27 Thread Tim Panton



On 27 Jun 2006, at 13:54, Il Neofita wrote:


Hi,
someone know a good webphone, possibily a free one

Thx



Ours isn't free - but take a look at
www.mexuar.com , or drop me an email.

Tim.

Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] isdn-data over iax

2006-06-27 Thread Florian Overkamp


[EMAIL PROTECTED] wrote:

is it possible to route an ISDN-Data channel over an iax-connection ?

the setup is 

pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk 
Server2 (E1)-connecting to an external isdn-dialin router


via the iax-line the call is transfered as speech which is not accepted at 
the remote end


IAX is not suited for this. Maybe TDMoE is an option for you ?

Florian
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Re: [Asterisk-Users] asterisk to mobile phone

what brand of gsm gateway do you think works well with asterisk?On 6/27/06, Colin Anderson [EMAIL PROTECTED]
 wrote:






A GSM 
gateway will allow you to specify a ruleset so a channel on the gateway is 
always locked to a particular mobile number, then you just send the call from 
Asterisk to the gateway and it will do the hunt for you. 

  -Original Message-From: Lito Lampitoc 
  [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 27, 2006 7:59 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] asterisk to mobile phoneIs it possible 
  to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 
  mobile phones brought by the engineers in the field, when someone calls the 
  trunkline, asterisk will hunt which of the 10 mobile phones is available. What 
  do I need for this setup? Thanks in 
advance.Lito

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RE: [Asterisk-Users] SE Michigan asterisk users group

2006-06-27 Thread Carlos Alperin

Jyran,

We didn't make any plans. Nobody still has confirm anything. All that I did
was a list of everyone that is interested.

I don't know where are we going to meet. If we 're going to meet. Or we may
be do a virtual meeting?

I only need to know if you are interested or not on participate.

Thanks, 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, June 27, 2006 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SE Michigan asterisk users group

I am in the Cadillac area and would be interested in joining this group
depending on where in the SE it is located and the time the meetings occur.

Thank you,

Jyran Glucky
Advisory Programmer
BlueWare, Inc.
Strategic HealthWare Solutions
3060 W. 13th Street
Cadillac, MI 49601
Phone:  (231) 779-0224 ext. 111
Fax: 231-779-1002
Skype: Jyran Glucky
AIM: JyranGlucky
mailto:[EMAIL PROTECTED]
http://www.blueware.net

BlueWare Fast Facts...
Interoperability Leader Recognized at HIMSS 2006 IBM Beacon Award Winner -
Best DB2 Document  Management Solution Worldwide Awarded 2006 Top 50 Growing
Companies in Michigan A Decade of Delivering the First Best-of-Breed EHR
Solution to the Market IBM Premier Business Partner


   
 Carlos Alperin  
 [EMAIL PROTECTED] 
 om.netTo 
 Sent by:  'Asterisk Users Mailing List - 
 asterisk-users-bo Non-Commercial Discussion' 
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 06/26/2006 11:58  RE: [Asterisk-Users] SE Michigan
 AMasterisk users group
   
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




Ok, I count at least 4. Just lets propose when  where for the first meeting
group, and start to think about issues discussion.

Tom, what to we need for the mailing list? I can do something about that.

Carlos Alperin



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Radon
Sent: Monday, June 26, 2006 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SE Michigan asterisk users group

I'm also in the area, near Southfield.  I'd be interested as well.

On 6/22/06, BerkHolz, Steven [EMAIL PROTECTED] wrote:
  I am thinking of getting an asterisk user group together for either SE
  Michigan or just Metro-Detroit.

  How much interest in asterisk in Michigan is there on this list?

  I am already on the board of glimasoutheast, with is a group for
  technology professionals. (very broad range)
  It is a spin-off from Automation Alley, which is SE Michigan's version
  of Silicone Valley.

  --
  Steven

  http://www.glimasoutheast.org

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Is it something someone said, was it something someone said?
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Re: [Asterisk-Users] Call length limitation

What does the CLI show when you make the call? That might help in diagnosingyour problem.
FlynnHi Flynn The situation looks like this:exten = _0800X.,1,AGI(/usr/share/asterisk/agi-bin/checklimit.php|${CALLERIDNUM}|${CONTEXT})exten = _0800X.,2,GotoIf($[${code} = 0 ]?0800-Calls|${EXTEN}|6:0800-Calls|${EXTEN}|3)
exten = _0800X.,3,GotoIf($[${code}  0  ${code}  20 ]?announcement|s|1:0800-Calls|${EXTEN}|4)exten = _0800X.,4,GotoIf($[${code} = 20 ]?announcement|s|4)exten = _0800X.,6,Dial(Sip/[EMAIL PROTECTED]
||L($[${limit}-1000]:6:3))exten = _0800X.,7,Hangup()exten = h,1,DeadAGI(/usr/share/asterisk/agi-bin/ending_call.php|${UNIQUEID}) -- SIP Seeding peer from astdb: '0327393001' at [EMAIL PROTECTED]
:5060 for 3600 -- Executing Goto(SIP/0327393001-ce12, 0800-Calls|0800102102|1) in new stack -- Goto (0800-Calls,0800102102,1) -- Executing AGI(SIP/0327393001-ce12, /usr/share/asterisk/agi-bin/checklimit.php|0327393001|0800-Calls) in new stack
 -- Launched AGI Script /usr/share/asterisk/agi-bin/checklimit.php -- AGI Script /usr/share/asterisk/agi-bin/checklimit.php completed, returning 0 -- Executing GotoIf(SIP/0327393001-ce12, 1?0800-Calls|0800102102|6:0800-Calls|0800102102|3) in new stack
 -- Goto (0800-Calls,0800102102,6) -- Executing Dial(SIP/0327393001-ce12, Sip/[EMAIL PROTECTED]||L(1:6:3)) in new stack -- Limit Data for this call: -- - timelimit = 1
 -- - play_warning = 0 -- - play_to_caller= yes -- - play_to_callee= no -- - warning_freq = 0 -- - start_sound = -- - warning_sound = timeleft -- - end_sound = beep
 -- Called [EMAIL PROTECTED] -- SIP/ipfon-2802 is ringing -- SIP/ipfon-2802 is making progress passing it to SIP/0327393001-ce12 -- SIP/ipfon-2802 answered SIP/0327393001-ce12 -- SIP Seeding peer from astdb: '0327393001' at 
[EMAIL PROTECTED]:5060 for 3600 After 10 sec I hear beep and: -- Playing 'beep' (language 'en') And that's it. The call is limitated but the h extension is not executed until I hang up manually.

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RE: [Asterisk-Users] Re: SE Michigan asterisk users group

2006-06-27 Thread Carlos Alperin

Thanks Steven, This is my first answer.

I'll going to make [EMAIL PROTECTED] the mailing list. I'm going to
include to everyone of the people that has exchange e-mail about this.

After that, we can take care of what to do in advance.

Carlos Alperin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Tuesday, June 27, 2006 9:41 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: SE Michigan asterisk users group

We should figure out what we are doing before we do any investing in domain
names, etc.

I believe that we have the option to be a sub-group off of the glima network
if desired.
http://www.glima.org/autoalley/GLIMA+Network/Member+Benefits/
They are a non-profit org for technology people in Michigan.
They are backed by Automation Alley and have resources if needed. (meeting
rooms, non-profit paperwork help, etc.) It may also help with membership, as
we would be able to post our meetings on their calendar.
http://www.glima.org/autoalley/GLIMA+Network/Calendar+of+Events/Calendar.htm
?ViewMonth=6-27-2006SubBrandId=0

Carlos,  If you would like to set up a temp. mailinglist, feel free. (this
will keep us from clogging this list.)

Please let me know if you guys think I should pursue starting the group as a
subgroup of glima.


--
--
Steven

http://www.glimasoutheast.org



Carlos Alperin [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
 OK,

 This is what I get till now:

 Ron Kushner Sterling Heigths [EMAIL PROTECTED] Rusty Dekema Ann Arbor 
 [EMAIL PROTECTED] Jon Radon Southfield [EMAIL PROTECTED] Steven 
 BerkHolz  [EMAIL PROTECTED] Tom Hayden Livonia 
 [EMAIL PROTECTED] Michael George Lansing [EMAIL PROTECTED] Tim 
 Sharp Livonia [EMAIL PROTECTED] Bradley Watkins ?? 
 [EMAIL PROTECTED] albeit ?? [EMAIL PROTECTED] Carlos 
 Alperin Southfield [EMAIL PROTECTED]

 If you guys want to start planning on how to start, then is time to 
 start exchanging direct mails with more info In order to arrange 
 future steps.

 I can create the mailing list, for that I can use my own domain 
 calperin.com, or we going to need to register a domain on Network 
 Solutions

 Waiting for suggestions,

 Regards,

 Carlos Alperin

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Re: [Asterisk-Users] asterisk to mobile phone

btw, i got it, 2N Easygate is highly compatible with Asterisk. Thanks.On 6/27/06, Lito Lampitoc [EMAIL PROTECTED]
 wrote:what brand of gsm gateway do you think works well with asterisk?
On 6/27/06, Colin Anderson 
[EMAIL PROTECTED]
 wrote:







A GSM 
gateway will allow you to specify a ruleset so a channel on the gateway is 
always locked to a particular mobile number, then you just send the call from 
Asterisk to the gateway and it will do the hunt for you. 

  -Original Message-From: Lito Lampitoc 
  [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 27, 2006 7:59 
  AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] asterisk to mobile phoneIs it possible 
  to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 
  mobile phones brought by the engineers in the field, when someone calls the 
  trunkline, asterisk will hunt which of the 10 mobile phones is available. What 
  do I need for this setup? Thanks in 
advance.Lito

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Re: [Asterisk-Users] Call length limitation

2006-06-27 Thread William Piper

Why not add the g parameter and make your deadAGI as the next priority?
I think that would accomplish what you are trying to do.

Example:
exten = x,1,Dial(Sip/|30|gL(6:3:1))exten = x,2,DeadAGI()bp

On 6/27/06, El Flynn [EMAIL PROTECTED] wrote:
Andrew Nowrot wrote: Hi I have a problem with Dial application. The dialplan looks like this:
 ; exten = x,1,Dial(Sip/|30|L(6:3:1)) exten = x,2,Hangup() exten = h,1,DadAGI() ; The call is limited to 60 sec and after that time the conversation stops,
 but Asterisk never reach theh extension. I could use the S() option in Dial application but I want to have the announcements. Is there any way to force Asterisk to execute the DeadAGI. Or
 maybe there is some other solution to achieve a goal.What does the CLI show when you make the call? That might help in diagnosingyour problem.Flynn___
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[Asterisk-Users] ExternalIVR vs AGI

2006-06-27 Thread Daniel Salama

I have an Perl AGI script that acts as an IVR for my Asterisk box.  
Basically, it simply plays audio files to the caller, collecting DTMF  
input and logging every DTMF input into a database table, simply to  
document every step or option selected by the caller.


One thing is that in addition to playing audio files, it also, at  
some point, plays SayUnitTime command.


This IVR has constantly about 20 simultaneous callers 24x7.

Would it be more resource efficient to migrate this to ExternalIVR?  
What are the pros/cons of using ExternalIVR vs using my Perl AGI.


Thanks,
Daniel
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RE: [Asterisk-Users] asterisk to mobile phone

2006-06-27 Thread Colin Anderson




I use 
an Ateus VoiceBlue which allows you to do this (never tried it though) which is 
a SIP device and you write your dialplan to send calls to the SIP device just 
like ringing an extension in Asterisk. It works fine but it tends to drop calls 
under load so I have an AGI that determines the load and if it goes beyond a 
certain threshold it relays the calls out the PSTN. By load, I mean as much as 
four calls hitting it simultaneously, if it only gets onsie-twsie calls it works 
fine. 

  -Original Message-From: Lito Lampitoc 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, June 27, 2006 9:03 
  AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] asterisk to mobile 
  phonewhat brand of gsm gateway do you think works well 
  with asterisk?
  On 6/27/06, Colin 
  Anderson [EMAIL PROTECTED] 
   wrote:
  


A GSM gateway will allow 
you to specify a ruleset so a channel on the gateway is always locked to a 
particular mobile number, then you just send the call from Asterisk to the 
gateway and it will do the hunt for you. 


  -Original 
  Message-From: Lito Lampitoc [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 27, 
  2006 7:59 AMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] asterisk to mobile phoneIs it 
  possible to trunk hunt mobile phones in asterisk? say I have one trunkline 
  and 10 mobile phones brought by the engineers in the field, when someone 
  calls the trunkline, asterisk will hunt which of the 10 mobile phones is 
  available. What do I need for this setup? Thanks in 
  advance.Lito
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RE: [Asterisk-Users] best hardphone for Asterisk?

2006-06-27 Thread shadowym

Which public STUN servers are you using or did you setup your own? 

 -Original Message-
 From: Cullin J. Wible [mailto:[EMAIL PROTECTED] 
 Sent: Monday, June 26, 2006 8:56 PM
 To: 'Asterisk Users Mailing List - Non-Commercial 
 Discussion'; 'Iain Barker'
 Subject: RE: [Asterisk-Users] best hardphone for Asterisk?

 We've used a number of the polycom 301 and 501 phones in our office.

 We have also deployed a dozen of the Linksys SPA-1001 
 single-line FXS adapters using G726, SIP, NAT and STUN. They 
 are extremely reliable and easy to deploy - $60-$70 US each.

 We tested a number of IAX hard phones and didn't find 
 anything that was reliable and/or suitable for our corporate 
 setting. We really wanted to run IAX for remote users, but 
 eventually decided that SIP/STUN was easier to support.

 We also tested the IAXy device and found that it's inability 
 to use DNS resolution, only be configured on Linux, and only 
 run ulaw/alaw made and that it cost more then the SPA-1001, 
 which can use DNS, G726/G729 and has web-based configuration 
 for less money the more attractive option.

 We also tested the IAX hard phone made by AT-COM only to find 
 that a number of features such as call transfer do not work.

 For home/remote users: setup STUN, and use a SPA-1001. For a 
 corporate setting I highly recommend the Polycom phones.

 Cheers,

 Cullin J. Wible
 Co-Founder  CTO
 Email Data Source, Inc.
 212-514-8900 x1006

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Doug Crompton
 Sent: Monday, June 26, 2006 11:49 PM
 To: Iain Barker
 Cc: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] best hardphone for Asterisk?

 Iain,

  Thanks for the repsonse but you are kidding me right? From 
 what I can see if I bought this phone and two remotes my 
 outlay would be close to $800 US.
 This is NOT a home device unless you have nothing better to 
 do with your money!

 You can buy a lot of single line wireless phones and FXS 
 devices for that amount!

 Doug

 On Mon, 26 Jun 2006, Iain Barker wrote:

  Doug,

  What you are describing sounds like the Aastra 480-CT, a base 
  Ethernet/SIP screenphone supporting multiple wireless 
 handsets [but as 
  this is a non-commercial list I won't go into more detail 
 here, google 
  for the above model number if you're interested in more info.]

  - Iain

  ---
  Message: 4
  Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT)
  From: Doug Crompton [EMAIL PROTECTED]
  Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Message-ID:
  [EMAIL PROTECTED]
  Content-Type: TEXT/PLAIN; charset=US-ASCII

  Still awfully pricey for home use and the styling is not 
 there for a 
  bedroom or many other areas of a modern home. What we need is a 
  wireless sip phone modeled like the panasonic or uniden which allow 
  multiple extension off of one base. The base would connect to the 
  internet. The other problem is many of these phones require 
 power, so 
  even if you have backup for your central system the phone 
 still needs 
  to be on it. Power over ethernet would help.

  Doug

 Those that sacrifice essential liberty to obtain a little 
 temporary safety deserve neither liberty nor safety.  -- Ben 
 Franklin (1759)

 *  Doug Crompton *
 *  Richboro, PA 18954*
 *  215-431-6307  *
 **
 * [EMAIL PROTECTED]*
 * http://www.crompton.com  *

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Re: [Asterisk-Users] Call length limitation

2006-06-27 Thread Michiel van Baak

On 11:18, Tue 27 Jun 06, William Piper wrote:
 Why not add the g parameter and make your deadAGI as the next priority?
 I think that would accomplish what you are trying to do.
 
 Example:
 exten = x,1,Dial(Sip/|30|gL(6:3:1))
 exten = x,2,DeadAGI()

Dont use DeadAGI on non-hungup channels.
So in this case it will have to be another AGI call.

-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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

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RE: [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed

2006-06-27 Thread Mark Adams









Hello, 

Anyone here
have experience with Audiocodes MediaPack MP-108 Gateways? 

I would
be willing to pay someone for advice and support with configuring my gateways
for a telemarketing project I am starting. My experience is somewhat limited
but all I want to do is make outbound calls just like I would on normal pots
lines. (Thats the best way to explain it) I do not need any special
configuration nor am I going to use it for any incoming calls. I would like to
just have the gateways register and properly send calls out and relay DTMF
tones. After I get them up and running I should have the manual read and
digested by then and I will be good to go. 



Anyone interested
please email me off list 

Mark
Adams 
Infinity Marketing
216-334-9304 


 
  
  
  
 
 
  
  
  
 









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[Asterisk-Users] Modifying Voicemail menus?

2006-06-27 Thread Dan Elder

Is there a way to edit the options available in the voicemail menu trees? My
users are complaining that it's too complicated (I know, it's not really
complicated), and I wanted to remove some of the options if this is
possible. So far I havent' found any info on the wiki or searches, not that
it isn't out there.. I just cant' seem to find it.. Any pointers?

Thanks

Dan

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Re: [Asterisk-Users] 7960 help: transferring calls

2006-06-27 Thread Lacy Moore - Aspendora

Blind transfer is not possible via softkeys on the 7960 using chan_sccp. However, check your features.conf. You should have a line in there regarding blind transfer. I believe the default is #, but it recommended changing that to ##. I did this, and on my 7960s, you hit ## then the extension you wish to transfer to. This does requires use of the t or T dial command options. You'll have to look this up, as I can't remember offhand which is needed, and it depends on whether the 7960 is the caller or callee.

#2 and #3 I'll have to check on. If I remember correctly, it's pretty easy.
On 6/27/06, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings all,Not specifically an asterisk query, but a couple of transfer queries thatI'm sure are obvious to folks who use these phones all the time:
1) how does one do a blind transfer? When a call is answered and one hitsthe transfer button, followed by an extension, one has to wait for the otherparty to answer, then hit transfer again, before the call is released. I'm
sure there must be an option to answer a call, then fire it straight off toanother extension without waiting for an answer?2) if there are 2 incoming calls currently on the go (i.e. the first one hasbeen put on hold for the operator to answer the second call), how does one
determine which call will be transferred when the transfer button ispressed?Is there a way to select the source call for a transfer prior tohitting transfer?3) when handling 2 calls, how does one swap between them?
These phones are running sccp through chan_sccp if that makes any differenceto operation.Thanks in advance folks.Regards,Chris--C.M. Bagnall, Director, Minotaur I.T. Limited
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RE: [Asterisk-Users] 7960 help: transferring calls

2006-06-27 Thread Ryan Amos

Chan_sccp does not support blind transfer. I would suggest using
chan_sip and the SIP images with these phones; it is much more stable,
has more features and is being actively developed. Chan_sip supports
blind transfer and 3-way calling, plus it handles multiple calls on hold
a bit more gracefully than chan_sccp. Chan_sccp seems largely dead at
this point; the maintainer has not released a patch in 2 months and most
of the users who know the code well enough to possibly maintain it seem
to have moved on to other projects.

If you don't have access to the SIP firmware (which you can get for a
$75 cisco smartNET contract,) understand that chan_sccp still has quite
a few bugs that make it unsuitable for a production system in my eyes.
There are still unresolved deadlocks and channel locking issues which
can render your phone unusable until an asterisk restart, and you can't
reload the configuration without unloading the driver and killing
registration on all your phones (meaning you can't add a phone without
downtime for the whole system.)

But this is just my take on the situation.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: Tuesday, June 27, 2006 9:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] 7960 help: transferring calls

Greetings all,

Not specifically an asterisk query, but a couple of transfer queries
that
I'm sure are obvious to folks who use these phones all the time:

1) how does one do a blind transfer? When a call is answered and one
hits
the transfer button, followed by an extension, one has to wait for the
other
party to answer, then hit transfer again, before the call is released.
I'm
sure there must be an option to answer a call, then fire it straight off
to
another extension without waiting for an answer?

2) if there are 2 incoming calls currently on the go (i.e. the first one
has
been put on hold for the operator to answer the second call), how does
one
determine which call will be transferred when the transfer button is
pressed?  Is there a way to select the source call for a transfer prior
to
hitting transfer?

3) when handling 2 calls, how does one swap between them?

These phones are running sccp through chan_sccp if that makes any
difference
to operation.

Thanks in advance folks.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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RE: [Asterisk-Users] Modifying Voicemail menus?

2006-06-27 Thread Cullin J. Wible

We did it by comment out a number of lines in the code and then re-compiled
just that module.

We also did the same for the company directory.

Other then that I'm not sure if there's much you can do.

Cullin J. Wible
Co-Founder  CTO
Email Data Source, Inc.
212-514-8900 x1006
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Tuesday, June 27, 2006 12:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Modifying Voicemail menus?

Is there a way to edit the options available in the voicemail menu trees? My
users are complaining that it's too complicated (I know, it's not really
complicated), and I wanted to remove some of the options if this is
possible. So far I havent' found any info on the wiki or searches, not that
it isn't out there.. I just cant' seem to find it.. Any pointers?

Thanks

Dan

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Re: [Asterisk-Users] Call length limitation

Thanks for all repliesI noticed that L option does not hangup the call it only limits the call. (In my case the h extension isn't executed). S option can do that (Asterisk reach the h extension)L(x:y:z) - do not hang up the call after x sec.
S(x) - hangup the call after x sec.I also noticed that when I combine those to (S and L) only L is working.Is there any way to use L option and have also the opportunity to hang up the call?
CheersAndrew
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Re: [Asterisk-Users] Call length limitation

2006-06-27 Thread William Piper

Although I've never tried it along withthe L option, you couldtry absolutetimeout:

exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))bp

On 6/27/06, Andrew Nowrot [EMAIL PROTECTED] wrote:

Thanks for all repliesI noticed that L option does not hangup the call it only limits the call. (In my case the h extension isn't executed). S option can do that (Asterisk reach the h extension)L(x:y:z) - do not hang up the call after x sec. 
S(x) - hangup the call after x sec.I also noticed that when I combine those to (S and L) only L is working.Is there any way to use L option and have also the opportunity to hang up the call?
Cheers
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[Asterisk-Users] Realtime Voicemail

I'm noticing that the documentation on the voip wiki for voicemail and realtime 
voicemail hasn't kept up with reality.

I just created a column called maxmsg in my table. I set it to 1 for the user. 
I can leave more than once voicemail message.
Why?

Doug.
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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 23, Issue 182

2006-06-27 Thread Dan Elder

We did it by comment out a number of lines in the code and then re-compiled
just that module.

Thx Cullin for the reply, has anyone made a flow chart or end user
instructions for comedian mail? Jus trying not to reinvent the wheel if it's
already been done.

Thanks!

Dan

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Re: [Asterisk-Users] Call length limitation

On 6/27/06, William Piper [EMAIL PROTECTED] wrote:
Although I've never tried it along withthe L option, you couldtry absolutetimeout:

exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))I didn't help still the same :(.
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RE: [Asterisk-Users] Realtime Voicemail

 -Original Message-
 From: Douglas Garstang 
 Sent: Tuesday, June 27, 2006 11:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Realtime Voicemail

 I'm noticing that the documentation on the voip wiki for 
 voicemail and realtime voicemail hasn't kept up with reality.

 I just created a column called maxmsg in my table. I set it 
 to 1 for the user. I can leave more than once voicemail message.
 Why?

Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogins for 
the user to 1, and it's letting me put the wrong pin in 3 times before 
disconnecting me. What am I missing here? Are the supported options documented 
somewhere, that matches up with what's really in the code?

Doug.
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Re: [Asterisk-Users] Realtime Voicemail

2006-06-27 Thread Michiel van Baak

On 12:13, Tue 27 Jun 06, Douglas Garstang wrote:
  -Original Message-
  From: Douglas Garstang 
  Sent: Tuesday, June 27, 2006 11:55 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] Realtime Voicemail

  I'm noticing that the documentation on the voip wiki for 
  voicemail and realtime voicemail hasn't kept up with reality.

  I just created a column called maxmsg in my table. I set it 
  to 1 for the user. I can leave more than once voicemail message.
  Why?

 Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogins for 
 the user to 1, and it's letting me put the wrong pin in 3 times before 
 disconnecting me. What am I missing here? Are the supported options 
 documented somewhere, that matches up with what's really in the code?

Do you cache realtime stuff ?
If so, that would explain it
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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

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Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000

2006-06-27 Thread Mike Lynchfield

HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's
seem siemens are made for europe style ring voltage not north american.On 6/27/06, Herchi Silviu 
[EMAIL PROTECTED] wrote:




Hello,

The main differences I can see:

- in zaptel.conf
you have span=1,0,0,ccs,hdb3, which means you ask Asterisk 
to serve as a timer for the PBX - on my setup the PBX is the master clock and 
Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use 
CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4)

- in zapata.conf
I have switchtype=EuroISDN. Generally speaking, try using 
other switchtypes.

Regards,

Silviu



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Josué 
ContiSent: 27 June 2006 14:41To: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: 
Asterisk x Siemens HiPath 4000

Silviu, 
thank's will be this attention. Below my configurations of zapata.conf and 
zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us
#zapata.conf
[trunkgroups]
[channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable 
= 
yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yes
cancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes 
rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31

Best Regards

Josué

2006/6/27, Herchi Silviu [EMAIL PROTECTED]: 


  
  
  Hi, 
  Could you post your /etc/zaptel.conf and 
  zapata.conf? 
  Also, is everything OK the other way round (i.e., 
  from the SIP phones to the PBX)? 
  Silviu 
   
  Hello 
  all. I have installed and functioning 
  asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is 
  interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, 
  the one that is happening he is that the calls originated for PABX Siemens and 
  destined to SIP phones asterisk are being without audio, nor Ring, is dumb. 
  They could help in this case me? 
  
  
  Best Regards  Josué 
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RE: [Asterisk-Users] Realtime Voicemail

 -Original Message-
 From: Michiel van Baak [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, June 27, 2006 12:21 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Realtime Voicemail

 On 12:13, Tue 27 Jun 06, Douglas Garstang wrote:
   -Original Message-
   From: Douglas Garstang 
   Sent: Tuesday, June 27, 2006 11:55 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: [Asterisk-Users] Realtime Voicemail

   I'm noticing that the documentation on the voip wiki for 
   voicemail and realtime voicemail hasn't kept up with reality.

   I just created a column called maxmsg in my table. I set it 
   to 1 for the user. I can leave more than once voicemail message.
   Why?

  Weird. Maxmsg suddenly worked on the next call. I tried 
 setting maxlogins for the user to 1, and it's letting me put 
 the wrong pin in 3 times before disconnecting me. What am I 
 missing here? Are the supported options documented somewhere, 
 that matches up with what's really in the code?

 Do you cache realtime stuff ?
 If so, that would explain it

I wasn't aware that realtime voicemail supported caching. I knew sip.conf did, 
but voicemail?
How does that work?

I just tried setting 'format' and 'sendvoicemail' in the users database row. No 
effect.
BUT... maxmsg DOES work... I don't understand.

Doug.
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Re: [Asterisk-Users] best hardphone for Asterisk?

2006-06-27 Thread Mike Lynchfield

We use cisco 7960's but thats not cheap..BTW Doungyour signature :Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety. -- Ben Franklin (1759)
is a good one.. tell that to your president..and the patriot act.s/patriot/cutallrights/PS Andrew.. penguins as in linux based ?or the phone just quacks all the time ?;)
On 6/27/06, shadowym [EMAIL PROTECTED] wrote:
Which public STUN servers are you using or did you setup your own? -Original Message- From: Cullin J. Wible [mailto:[EMAIL PROTECTED]] Sent: Monday, June 26, 2006 8:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'Iain Barker' Subject: RE: [Asterisk-Users] best hardphone for Asterisk? We've used a number of the polycom 301 and 501 phones in our office.
We have also deployed a dozen of the Linksys SPA-1001 single-line FXS adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy to deploy - $60-$70 US each. We tested a number of IAX hard phones and didn't find
anything that was reliable and/or suitable for our corporate setting. We really wanted to run IAX for remote users, but eventually decided that SIP/STUN was easier to support. We also tested the IAXy device and found that it's inability
to use DNS resolution, only be configured on Linux, and only run ulaw/alaw made and that it cost more then the SPA-1001, which can use DNS, G726/G729 and has web-based configuration for less money the more attractive option.
We also tested the IAX hard phone made by AT-COM only to find that a number of features such as call transfer do not work. For home/remote users: setup STUN, and use a SPA-1001. For a
corporate setting I highly recommend the Polycom phones. Cheers, Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006
-Original Message- From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Doug Crompton Sent: Monday, June 26, 2006 11:49 PM To: Iain Barker Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] best hardphone for Asterisk? Iain,Thanks for the repsonse but you are kidding me right? From what I can see if I bought this phone and two remotes my
outlay would be close to $800 US. This is NOT a home device unless you have nothing better to do with your money! You can buy a lot of single line wireless phones and FXS devices for that amount!
Doug On Mon, 26 Jun 2006, Iain Barker wrote: Doug, What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP screenphone supporting multiple wireless
handsets [but as this is a non-commercial list I won't go into more detail here, google for the above model number if you're interested in more info.] - Iain
--- Message: 4 Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT) From: Doug Crompton [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] best hardphone for Asterisk?
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID:
[EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Still awfully pricey for home use and the styling is not
there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the
internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help.
Doug Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety.-- Ben Franklin (1759)
*Doug Crompton * *Richboro, PA 18954* *215-431-6307* ** *
[EMAIL PROTECTED]* * http://www.crompton.com* ___ --Bandwidth and Colocation provided by
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[Asterisk-Users] PRI - Ring requested on channel errors - inbound outbound stop working.

2006-06-27 Thread Dan Sully


A few days ago, I started getting these errors on my Asterisk (1.2.9.1) console:

   -- Executing Queue(Zap/1-1, sales|tT|||3600) in new stack
   -- Channel 0/2, span 1 got hangup
   -- Channel 0/1, span 1 got hangup request
Jun 27 10:53:27 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:29 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:31 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:33 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:35 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:36 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:38 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
   -- Channel 0/1, span 1 got hangup
Jun 27 10:53:46 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jun 27 10:53:49 WARNING[18863]: chan_zap.c:8518 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.

At this point, any inbound and outbound calls fail. If I restart asterisk,
things work fine for a few days until it happens again.

Any thoughts on this issue?

Running zaptel 1.2.6 with libpri 1.2.3 and a TE110P, with the wcte11xp driver 
on Linux.

Thanks.

-D
--
Daemon seriously, first there was the circle, then sliced bread, then tivo
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Re: [Asterisk-Users] Polycom Buddies in 1.6.6

2006-06-27 Thread Ryan Stark

So I've got a 601 (1.6.6) with the side car, and the buddy watch seems to be working but it updates the statuses unreliably. When I do a sip show subscriptions in asterisk it lists my phone 12 times and at the bottom it says 0 active SIP subscriptions(s) I've got an older CVS-HEAD build, pre
1.2, do you think my problems are polycom or asterisk based?-RyanOn 6/19/06, Kevin P. Fleming [EMAIL PROTECTED]
wrote:- Douglas Garstang
[EMAIL PROTECTED] wrote: Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie
send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work?
Yes, it works on the Polycom 601 on my desk. However, the release notes say that the restriction was only removed for the IP600 and IP601; if you are using an IP300/1, IP500/1 or IP430 than the 7 buddy limit will still be in effect.
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RE: [Asterisk-Users] Realtime Voicemail

 -Original Message-
 From: Douglas Garstang 
 Sent: Tuesday, June 27, 2006 12:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Realtime Voicemail

  -Original Message-
  From: Douglas Garstang 
  Sent: Tuesday, June 27, 2006 11:55 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] Realtime Voicemail

  I'm noticing that the documentation on the voip wiki for 
  voicemail and realtime voicemail hasn't kept up with reality.

  I just created a column called maxmsg in my table. I set it 
  to 1 for the user. I can leave more than once voicemail message.
  Why?

 Weird. Maxmsg suddenly worked on the next call. I tried 
 setting maxlogins for the user to 1, and it's letting me put 
 the wrong pin in 3 times before disconnecting me. What am I 
 missing here? Are the supported options documented somewhere, 
 that matches up with what's really in the code?

Oh man, this is some freaky stuff. I commented out 'format=wav49|gsm|wav' in 
voicemail.conf and did a reload.
I set the format field to the user to 'gsm'.
And... Asterisk record a wav file. Huh?

Doug.

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Re: [Asterisk-Users] Realtime Voicemail

2006-06-27 Thread Michiel van Baak

On 12:26, Tue 27 Jun 06, Douglas Garstang wrote:
 I wasn't aware that realtime voicemail supported caching. I knew sip.conf 
 did, but voicemail?
 How does that work?
 
 I just tried setting 'format' and 'sendvoicemail' in the users database row. 
 No effect.
 BUT... maxmsg DOES work... I don't understand.

Looks like I was reading with my eyes closed.
I dont recal realtime voicemail using caching. Sorry.

Looks like you found an issue here. Sorry I cannot help you.
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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

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Re: SV: [Asterisk-Users] Error in config sample for GoToIf?

2006-06-27 Thread Mike Lynchfield

BLAH=1BLAH=1On 6/27/06, Brian Capouch [EMAIL PROTECTED] wrote:
Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write
 Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1Through more testing, the double quotes I used seemed superfluous; if
you use them in both places, or in neither, it works the same.But your example above lacks the $ ahead of the left brace.It is*that* which I now believe is in error in the example.Plus there seems to be confusion, on the Wiki at least, as to what
values mean what for ${AVAILSTATUS}Thx.B.--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___
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RE: [Asterisk-Users] Polycom Buddies in 1.6.6