date:20051230

On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
 Hi,

 I've just received a brand new td2400e , Where i can found some
 documentation for this card?, Digium's site do not show very usefull.
 I'd like to know how to configure zaptel.conf and zapata.conf  basically.

 Thanks, and Happy New Year to all.


 What kind of modules did you get with the card? FXO or FXS?


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RE : [Asterisk-Users] Howto config tdm2400

2005-12-30 Thread f6hqz-m

Hello,

Do as with a TDM400P, but use the correct driver (modprobe wctdm24xxp).
You have only more channels, it's all !
Insert the quad modules starting from number 1 printed place on the PCB.
This card run well and echocancel is very good.

Good luck !

Francois BERGERET,
[EMAIL PROTECTED],
France.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Manuel Casal
Envoyé : vendredi 30 décembre 2005 13:06
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [Asterisk-Users] Howto config tdm2400


Hi,

I've just received a brand new td2400e , Where i can found some 
documentation for this card?, Digium's site do not show very usefull.  
I'd like to know how to configure zaptel.conf and zapata.conf  basically.

Thanks, and Happy New Year to all.

-- 
Manuel Casal
[EMAIL PROTECTED]

[EMAIL PROTECTED]
Sistemas de Información y Protección de Datos, S.L.
Telf. + 34 902 678006
e-mail: [EMAIL PROTECTED]
web: http://www.e-sistemas.net


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Re: [Asterisk-Users] Semi-OT: porting numbers away


  Anyone have any info on porting numbers away from a VoIP provider to a Ma
  Bell or the like?  Thanks!!
 
 I had a friend port his from Bell -VOIP -VOIP.  He had no trouble.
 
 I would use a couple providers.  So this way if one goes down there is a 
 backup.

In very general terms (at least in the US), telephone numbers that are
considered portable can be moved from one itsp to another. However, the
move process generally involves a request for that move on the part of
the receiving itsp and an acknowledgement on the part of the old itsp
(or original owner of the number).

That transfer process has had lots of problems of which some include:
- some itsp's don't have a clue how to do it
- some telco's refuse to acknowledge the transfer
- etc, etc.

Most of the larger telco's will handle such requests rather well.


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Re: [Asterisk-Users] Howto config tdm2400

2005-12-30 Thread Manuel Casal


BJ Weschke escribió:

On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
  

Hi,

I've just received a brand new td2400e , Where i can found some
documentation for this card?, Digium's site do not show very usefull.
I'd like to know how to configure zaptel.conf and zapata.conf  basically.

Thanks, and Happy New Year to all.




 What kind of modules did you get with the card? FXO or FXS?


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I have a tdm2403e with 3 fxo modules plus echo cancelation. But in the 
future u must add a fsx module so id like to learn to configure both...



--
Manuel Casal
[EMAIL PROTECTED]

[EMAIL PROTECTED]
Sistemas de Información y Protección de Datos, S.L.
Telf. + 34 902 678006
e-mail: [EMAIL PROTECTED]
web: http://www.e-sistemas.net



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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Austin Denyer


On Fri, 30 Dec 2005 07:23:30 -0600
Rich Adamson  [EMAIL PROTECTED] wrote:
 
 In very general terms (at least in the US), telephone numbers that are
 considered portable can be moved from one itsp to another. However,
 the move process generally involves a request for that move on the
 part of the receiving itsp and an acknowledgement on the part of
 the old itsp (or original owner of the number).
 
 That transfer process has had lots of problems of which some include:
 - some itsp's don't have a clue how to do it
 - some telco's refuse to acknowledge the transfer
 - etc, etc.
 
 Most of the larger telco's will handle such requests rather well.

The 'rules' are somewhat flaky though, as there is no legal requirement
for a provider to port within it's own network.  Whilst this may seem
pointless, consider the following:

You have a Boost Mobile pre-paid phone, and want to go to a Nextel
contract (or vice-versa).  Boost Mobile is owned by Sprint/Nextel, so
they are under no legal obligation to port the number (and believe me,
they won't).

However, if you got a Cingular pre-paid, you could port your Boost
Mobile number to Cingular, then port it from Cingular to Nextel.  Or,
you could go Nextel - Cingular - Boost Mobile.

Go figure.

Regards,
Ozz.


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[Asterisk-Users] Queue features

2005-12-30 Thread Dov Bigio




Hi,
I am using the Queue application for 5 queues I have in my Call Center, 
and will by the end of January, implement it for the rest of the company 
(another 10 queues).

One of the main problems I face and my call center managers are worried 
about is the fact that when an agent uses the DND button of the Softphone, call 
center managers have no way of monitoring this.

Is there a way to track this?

Thank you
Dov
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Re: [Asterisk-Users] Queue features

2005-12-30 Thread Michiel van Baak

On 11:38, Fri 30 Dec 05, Dov Bigio wrote:
 Hi,
 
 I am using the Queue application for 5 queues I have in my Call Center, and 
 will by the end of January, implement it for the rest of the company (another 
 10 queues).
 
 One of the main problems I face and my call center managers are worried about 
 is the fact that when an agent uses the DND button of the Softphone, call 
 center managers have no way of monitoring this.
 
 Is there a way to track this?
 

Hi,

I don't think so, this is a client side setting and the
phone will reply with a REDIRECT  sip messages when a call
is sent to the client. (if you use sip clients that is)

I use some cisco phones with chan_sccp and they store the
dnd setting in the astdb, so if you use those you can
monitor it with the cli command 'database show SCCP'
Here's an example:

2 phones not set to dnd:
sin*CLI database show SCCP
/SCCP/SEP0012D9166A2C : dnd=0,cfwdall=,cfwdbusy= 
/SCCP/SEP0015626A4B99 : dnd=0,cfwdall=,cfwdbusy= 

1 phone set to dnd, the other not:
sin*CLI database show SCCP
/SCCP/SEP0012D9166A2C : dnd=0,cfwdall=,cfwdbusy= 
/SCCP/SEP0015626A4B99 : dnd=1,cfwdall=,cfwdbusy= 

hope this helps


-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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

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Re: [Asterisk-Users] SetAccount missing?

2005-12-30 Thread Andrew Latham

RTFM

ns2*CLI show application SetAccount
ns2*CLI
  -= Info about application 'SetAccount' =-

[Synopsis]
Set the CDR Account Code

[Description]
  SetAccount([account]): This application will set the channel account code for
billing purposes.
  SetAccount has been deprecated in favor of the Set(CDR(accountcode)=account).



On 12/29/05, William M. Sandiford [EMAIL PROTECTED] wrote:

 I just upgraded my system to the latest svn-trunk

 I previously made extensive use of the SetAccount() function, but now I'm
 getting the following error

 Dec 29 20:54:08 WARNING[4925]: pbx.c:1679 pbx_extension_helper: No
 application 'SetAccount' for extension (voipsubscriber-in, x, 100)

 Has this function been deprecated?  If so, what method is used to replace
 its functionality.  I have noticed a lot of deprecated features, variables,
 etc, and the wiki usually explains that the application / variable is
 deprecated and what to use in its replacement.  The wiki entry for set
 account doesn't say anything

 http://www.voip-info.org/wiki-Asterisk+cmd+SetAccount

 Any Ideas, as you can see...its missing

 sip1*CLI show application set
 SetSetAMAFlagsSetCallerID
 SetCallerPres  SetCDRUserFieldSetCIDName
 SetCIDNum  SetGlobalVar   SetGroup
 SetMusicOnHold SetRDNIS   SetTransferCapability
 sip1*CLI show application Set

 Regards,
 Bill

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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
---
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Re: [Asterisk-Users] Problem getting D channel up on Sangoma A102


   I am installing an Asterisk box equipped with the Sangoma A102 card. The 
 telco
 just tested the PRI interface and it is ll ok. I
 now connect my Asterisk box and I can't get the D-Channel up. If I enable
 intense pri debug I see messages like the following:
 
 --SNIP START--
  [ 02 01 7f ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
  0 bytes of data
 -- Got SABME from network peer.
 Sending Unnumbered Acknowledgement
 
 
  [ 02 01 73 ]
 
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
  0 bytes of data
 
 -- Restarting T203 counter
 -- Restarting T203 counter
   == Primary D-Channel on span 1 up
 pbx*CLI
  [ 02 01 7f ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 extended) ]
  0 bytes of data
 -- Got SABME from network peer.
 Sending Unnumbered Acknowledgement
 
 
  [ 02 01 73 ]
 
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
  0 bytes of data
 
 -- Restarting T203 counter
 -- Restarting T203 counter
   == Primary D-Channel on span 1 up
 T203 counter expired, sending RR and scheduling T203 again
 Sending Receiver Ready (0)
 
 
  [ 00 01 01 01 ]
 
 
  Supervisory frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
  N(R): 000 P/F: 1
  0 bytes of data
 
 -- Restarting T203 counter
 -- Retrying poll with f-bit
 Sending Receiver Ready (0)
 
 
  [ 00 01 01 01 ]
 
 
  Supervisory frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
  N(R): 000 P/F: 1
  0 bytes of data
 
 -- Restarting T203 counter
 Stopping T_203 timer
 T_200 timer already going (3)
 
  Protocol Discriminator: Q.931 (8)  len=13
  Call Ref: len= 2 (reference 0/0x0) (Originator)
  Message type: RESTART (70)
  [18 03 a9 83 86]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
 
 Dchan: 0
 
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel
 
 Type: 3
 
Ext: 1  Channel: 6 ]
  [79 01 80]
  Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
 
 Channel (0) ]
 -- T200 counter expired, What to do...
 -- Retransmitting 17 bytes
 
 
  [ 00 01 00 01 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]
 
 
  Informational frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
  N(S): 000   0: 0
  N(R): 000   P: 1
  13 bytes of data
 
 -- Rescheduling retransmission (2)
 -- T200 counter expired, What to do...
 -- Timeout occured, restarting PRI
 Sending Set Asynchronous Balanced Mode Extended
 
 
  [ 00 01 7f ]
 
 
  Unnumbered frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
 
 extended) ]
 
  0 bytes of data
 
   == Primary D-Channel on span 1 down
 
 --SNIP END--
 
 
 Config is the following:
 
 zaptel.conf:
 span=1,1,2,esf,b8zs
 bchan=1-23
 dchan=24
 loadzone = us
 defaultzone=us
 
 zapata.conf
 [channels]
 language=fr
 context=from-pstn
 switchtype=national
 resetinterval=never
 signalling=pri_cpe
 faxdetect=incoming
 usecallerid=yes
 echocancel=yes
 echocancelwhenbridged=no
 echotraining=800
 group=1
 channel=1-23
 
 Any hints appreciated

A couple of items to check

How sure are you the d-channel is on channel 24?

Check with the telco when you are ready to test as they will typically
disable the pri link to reduce the number of alarms they receive. They
are running under the assumption that you've not installed your stuff
yet and won't activate the d-channel (etc) until they know you are truly
ready to test/use the circuit.

Ensure you've connected your Sangoma card to the telco jack using a
T1/E1 cable.

Last, it is not uncommon (at least in the US) to find newly installed
circuits that some telco technician left in loopback. In very general
terms, one of their last installation steps is for them to loop back the 
circuit (at your location) and then send/receive data through the circuit
from their central office to measure bit error rates, etc. If they forget 
to dump that loopback, you are essentially sending calls to a unterminated 
T1/E1 circuit.


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Re: [Asterisk-Users] Queue features

2005-12-30 Thread Giovanni Miano

You can check status of Peer with Asterisk Management  Interface (AMI)
www.voip-info.org/wiki-Asterisk+manager+API
Cheers,Giovanni Miano
2005/12/30, Dov Bigio [EMAIL PROTECTED]:







Hi,
I am using the Queue application for 5 queues I have in my Call Center, 
and will by the end of January, implement it for the rest of the company 
(another 10 queues).

One of the main problems I face and my call center managers are worried 
about is the fact that when an agent uses the DND button of the Softphone, call 
center managers have no way of monitoring this.

Is there a way to track this?

Thank you
Dov

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-- Giovanni Miano
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Re: [Asterisk-Users] Howto config tdm2400

On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
 
 I have a tdm2403e with 3 fxo modules plus echo cancelation. But in the
 future u must add a fsx module so id like to learn to configure both...


 Ok. So in /etc/zaptel.conf you add:

fxsks=13-24

 And in /etc/asterisk/zapata.conf you add:

group=0
context=whatever context you want inbound calls to go to
signalling=fxs_ks
channel = 13-24

 When you add in the FXS modules, you just change fxsks to fxoks and
fxo_ks respectively.

 More detailed documentation is available here:

 http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
 http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf



--
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RE: [Asterisk-Users] Can we dial agents from extensions.conf

2005-12-30 Thread vivek

Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-

exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup

where agent 12 is configured as :-

agent = 12,12, vivek

After the agent is logged in on extension no12 as follows
Callback Agent '12' logged in on 12

I try to dial 12 from another sip phone and get this:-
-- Executing Dial(SIP/62-c24e, Agent/12) in new stack
-- outgoing agentcall, to agent '12', on 'Local/[EMAIL PROTECTED],1'
-- Called 12
-- Executing Dial(Local/[EMAIL PROTECTED],2, Agent/12) in new stack
Dec 30 14:26:54 NOTICE[13289]: app_dial.c:1011 dial_exec_full: Unable to create 
channel of type 'Agent' (cause 17 - User busy)
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
  == Spawn extension (default, 12, 2) exited non-zero on 'Local/[EMAIL 
PROTECTED],2'
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'Local/[EMAIL 
PROTECTED],2'
  == No one is available to answer at this time (1:0/0/0)
-- Executing Hangup(SIP/62-c24e, ) in new stack
  == Spawn extension (inoffice, 12, 2) exited non-zero on 'SIP/62-c24e'
-- Executing Hangup(SIP/62-c24e, ) in new stack
  == Spawn extension (inoffice, h, 1) exited non-zero on 'SIP/62-c24e'


I am unable to figure out why it is happening like this. They are all in the 
same context. Also, the agent is not busy. Also, I wonder why it says Unable 
to creat0e chanel of type 'Agent' cause user busy.
Do you have any idea why is it happening so?
I tried to tweak in but was not successful. 


With warm regards.

Vivek J. Joshi.

[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.

--Optimism is a mania for saying things are well when one is in hell.



Alexander Lopez wrote:
 There are options for queues.conf to not allow callers to join a queue
 if no members are logged in, also you can 'call an agent' with the agent
 channel, (IE agent/100)
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  [EMAIL PROTECTED]
  Sent: Friday, December 30, 2005 7:17 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Can we dial agents from extensions.conf 
  
  Hello friends,
 I wanted to ask if we can dial agents like the way we dial 
  extensions. I wanted to try this because the  users can login 
  and others can dial them. If a person has not logged in, he 
  isnt avalaible. I dont want to put people in a queue. Has 
  anyone tried this before? I was trying to do it but was unsuccessful. 
  
  Please tell me if there is a tweak or a workaround for this. 
  
  
  With warm regards.
  
  Vivek J. Joshi.
  
  [EMAIL PROTECTED]
  Trikon electronics Pvt. Ltd.
  
  --Optimism is a mania for saying things are well when one is in hell.
  
  `
  

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[Asterisk-Users] Problem on ZAP channel

2005-12-30 Thread rbrahmbhatt

Hello group members,
This is my first mail to this list. I am having one problem. When I dial a
number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?

Thanks


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[Asterisk-Users] Regular Crashes

2005-12-30 Thread Andrew Gough



I have just setup asterisk on a debian sarge box. I am 
running Asterisk1.21 with AMP and chan_capi_cm 0.6.1 using a BT 
Speedway (AVM Fritz)ISDN card, connected to a BT ISDN2e line. Currently we 
have 6 extensions(SIP) configured all using CounterPath(Xten) eyebeam 
softphone.After many hours of Googling I have finally got it all setup 
andworking. We can transfer calls internally and make and receive 
externalcalls. Its all great except for stability 
issues!!Essentially every now and again, asterisk simply dies (2-3 
times aday). No warning, no error, just my console session outputs 
adisconnected from console message.Sometimes the crashes happen when 
you are on a call, other times whenthere is no-one in the office.The 
server is a brand new AMD 3400+ with 512Mb RAM. The other issueexperienced 
is occasional break up on inbound sound quality.Below are traces of the 
last two crashesAny Help much appreciatedRegardsAndrew 
GoughFIRST TRACE#0 0x400268b7 in pthread_mutex_trylock () 
from /lib/tls/libpthread.so.0No symbol table info available.#1 
0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at lock.h:597No 
locals.#2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at 
channel.c:671 f = {frametype = 4, 
subclass = 1, datalen = 0, samples = 0, mallocd = 0, offset = 0, src = 
"" data = "" delivery = {tv_sec =0, tv_usec = 0}, 
prev = 0x0, next = 0x0}#3 0x408fc2d9 in __sip_autodestruct 
(data="" at 
chan_sip.c:1315 p = (struct 
sip_pvt *) 0x81be208#4 0x08056c3e in ast_sched_runq (con=0x8172f28) at 
sched.c:373 current = (struct 
sched *) 0x8174868 tv = {tv_sec = 
1135275568, tv_usec = 989877} x = 
0 res = 1083432672#5 
0x40927e28 in do_monitor (data="" at 
chan_sip.c:11253 res = 
0 sip = (struct sip_pvt *) 
0x0 peer = (struct sip_peer *) 
0x0 t = 
1135275568 fastrestart = 
0 lastpeernum = 
-1 curpeernum = 
6 reloading = 0#6 
0x40024b63 in start_thread () from /lib/tls/libpthread.so.0No symbol table 
info available.#7 0x401ac18a in clone () from /lib/tls/libc.so.6No 
symbol table info available.SECOND TRACE#0 0x400268b7 
in pthread_mutex_trylock () from /lib/tls/libpthread.so.0No symbol table 
info available.#1 0x0806c146 in ast_mutex_trylock (pmutex=0x120010c) 
at lock.h:597No locals.#2 0x0806175a in ast_queue_hangup 
(chan=0x1200040) at channel.c:671 
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,mallocd = 0, 
offset = 0, src = ""> data = "" delivery = {tv_sec = 0, tv_usec = 
0}, prev = 0x0, next =0x0}#3 0x408fc2d9 in __sip_autodestruct 
(data="" at 
chan_sip.c:1315 p = (struct 
sip_pvt *) 0x81eb518#4 0x08056c3e in ast_sched_runq (con=0x8172f78) at 
sched.c:373 current = (struct 
sched *) 0x8174528 tv = {tv_sec = 
1135343875, tv_usec = 693503} x = 
1 res = 0#5 0x40927e28 
in do_monitor (data="" at 
chan_sip.c:11253 res = 
0 sip = (struct sip_pvt *) 
0x0 peer = (struct sip_peer *) 
0x0 t = 
1135343875 fastrestart = 
0 lastpeernum = 
-1 curpeernum = 
6 reloading = 0#6 
0x40024b63 in start_thread () from /lib/tls/libpthread.so.0No symbol table 
info available.#7 0x401ac18a in clone () from /lib/tls/libc.so.6No 
symbol table info available.___
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Re: [Asterisk-Users] Problem getting D channel up on Sangoma A102

2005-12-30 Thread David Yat Sin

Try to recompile/reinstall (make clean; make install) zaptel after the
wanpipe installation to have the new 'patched' zaptel modules installed on
your system. 

David Yat Sin
Sangoma Technologies
(905) 474-1990 x119
(800) 388-2475 x119
Fax: (905) 474 9223
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Website: www.sangoma.com
 

Message: 10
Date: Thu, 29 Dec 2005 13:35:19 -0500
From: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem getting D channel up on Sangoma A102
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

Hi all,

  I am installing an Asterisk box equipped with the Sangoma A102 card. The
telco just tested the PRI interface and it is ll ok. I now connect my
Asterisk box and I can't get the D-Channel up. If I enable intense pri debug
I see messages like the following:

--SNIP START--
 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]
 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement


 [ 02 01 73 ]


 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
 0 bytes of data

-- Restarting T203 counter
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
pbx*CLI
 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]
 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement


 [ 02 01 73 ]


 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
 0 bytes of data

-- Restarting T203 counter
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
T203 counter expired, sending RR and scheduling T203 again Sending Receiver
Ready (0)


 [ 00 01 01 01 ]


 Supervisory frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
 Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
 N(R): 000 P/F: 1
 0 bytes of data

-- Restarting T203 counter
-- Retrying poll with f-bit
Sending Receiver Ready (0)


 [ 00 01 01 01 ]


 Supervisory frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
 Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
 N(R): 000 P/F: 1
 0 bytes of data

-- Restarting T203 counter
Stopping T_203 timer
T_200 timer already going (3)

 Protocol Discriminator: Q.931 (8)  len=13 Call Ref: len= 2 (reference 
 0/0x0) (Originator) Message type: RESTART (70)
 [18 03 a9 83 86]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
 Exclusive

Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel

Type: 3

   Ext: 1  Channel: 6 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated

Channel (0) ]
-- T200 counter expired, What to do...
-- Retransmitting 17 bytes


 [ 00 01 00 01 08 02 00 00 46 18 03 a9 83 86 79 01 80 ]


 Informational frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
 N(S): 000   0: 0
 N(R): 000   P: 1
 13 bytes of data

-- Rescheduling retransmission (2)
-- T200 counter expired, What to do...
-- Timeout occured, restarting PRI
Sending Set Asynchronous Balanced Mode Extended


 [ 00 01 7f ]


 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode

extended) ]

 0 bytes of data

  == Primary D-Channel on span 1 down

--SNIP END--


Config is the following:

zaptel.conf:
span=1,1,2,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us

zapata.conf
[channels]
language=fr
context=from-pstn
switchtype=national
resetinterval=never
signalling=pri_cpe
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=1
channel=1-23

Any hints appreciated


Andre



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Re: [Asterisk-Users] Problem on ZAP channel

2005-12-30 Thread Michael Sampson

I'm not sure if I'm right about this. But I think with a regular phone 
connection. You first dial the number and send the digits to the PBX and 
than the PBX has to redial the digits on the real phone line. Hence the 
delay. I think you get that with all PBXs when dialing an outside line.


Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000



[EMAIL PROTECTED] wrote:


Hello group members,
This is my first mail to this list. I am having one problem. When I dial a
number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?

Thanks


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[Asterisk-Users] TDM400 FXO outbound issue

2005-12-30 Thread Jason D. Wolfe

Hello,

I'm rather new to Asterisk so I'm in the wrong group for this issue please
let me know.
I'm using TDM400p with 2 FXO's on channel 3-4, Fedora Core4, 2.6, udev, and
I have the card on it's own IRQ

when I drop a .call file into /var/spool/asterisk/outgoing I see the
following on the CLI...

--Attempting to call on Zap/4/phonenumber for [EMAIL PROTECTED]:1 (Retry
1)(where phonenumber is a 10 digit local number)
--Hungup 'Zap/4-1'
NOTICE[32626]: pbx_spool.c:270 attempt_thread: Call failed to go through,
reason 1

the phonenumber I am dialing does ring, but when I answer there is silence,
and when I hangup I see the second and third line above (NOTICE[32626].  It
seems like the Asterisk or the card isn't recognizing when I answer the
phone.

1. How can I get more debugging information out of the CLI, or other files
for that matter?
2. Any direction will be helpful as I feel that I've hit a wall... I've been
googling for 3 days and reading everything I can find.

If a simple soultion doesn't present itself soon I'm willing to pay for a
more detailed support as the main functional requirement of my system is to
make 2 outbound calls, do some processing and bridge the calls together.

Jason Wolfe
[EMAIL PROTECTED]
c (770) 561-6956

This e-mail transmission may contain information that is proprietary,
privileged and/or confidential and is intended exclusively for the person(s)
to whom it is addressed. Any use, copying, retention or disclosure by any
person other than the intended recipient or the intended recipient's
designees is strictly prohibited. If you are not the intended recipient or
their designee, please notify the sender immediately by return e-mail and
delete all copies.

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[Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?

2005-12-30 Thread Ryan Booz









Hey all



I have a Gentoo system with Asterisk 1.2 installed. Its
been working great, for some reason the Zaptel module for my Wildcard TDM
(wctdm) seems to go missing anytime the server reboots, causing me to have to
go to the Zaptel source directory and do a quick make install. This
is the first Linux box Ive administered in a number of years Gentoos
module stuff is a bit unfamiliar to me. Any idea what file is getting read at
boot thats taking wctdm out of the modprobe
path?



Any help on how to solve the problem would be much
appreciated. As it stands now, anytime I have to reboot the server, I have to
manually login, install the module and then start Asterisk.



Thanks!



Ryan Booz

Director of IT

Good Steward Software, LLC

111 Sowers Street, Suite 400

State College, PA 16801

Phone: 877-327-3702 x.26 (814-237-3744 x.26)

Fax: 719-623-0577

Visit us at www.energycap.com








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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Julio Arruda



Since the last hurricane (that left me without phone for around 3 weeks 
or so), I did the call forwarding (remote call forwarding in fact).

Lucky I was running in the cable modem in a couple of days (power restored).
I was planning in having two DIDs in distinct providers (I've been using 
them for outbound in BYOD contracts), and keep the POTS number in the 
more stable one.

But I'm still concerned with the 911 issue.
Is the POTS telco (Bellsouth in South Florida in my case) mandated to 
provide 911 in a ported line ? I'm not that confident in using 911 via 
the ITSP.



Kerry Garrison wrote:

-- Personal opinion alert --
 
Do not route everything to an ITSP. At minimum keep a main PSTN line with

call forwarding or call forwarding on busy until you are 1% confident
that the service works, is reliable, stable, and will have some staying
power.
 
Kerry Garrison

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

  _  


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Thursday, December 29, 2005 5:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Semi-OT: porting numbers away



I'm looking to move one of my clients to an Asterisk system and a VoIP
provider (Teliax, Voxee, ViaTalk, Voicepulse).  My concern is porting my
client's numbers to a VoIP provider.  Let's say we get all their numbers
ported to Teliax (or Voxee or viatalk, etc.), everything is peachy for a
year, then Teliax gets sued for some reason or another, and goes bankrupt
and closes its doors.  That, obviously, leaves my clients without phone
service.but what happens to their numbers?  If the VoIP provider goes out of
business, can I go to another VoIP provider or a ma bell and transfer the
numbers to them even if Teliax (or whomever) is unreachable and off the map?

 


Thanks in advance for any info!

 

 


-Ross






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Re: [Asterisk-Users] Queue features

2005-12-30 Thread Dov Bigio




But a peer whose Softphone is on DND mode is still 
considered available, isn't it?


  - Original Message - 
  From: 
  Giovanni 
  Miano 
  To: Dov Bigio ; Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, December 30, 2005 12:21 
  PM
  Subject: Re: [Asterisk-Users] Queue 
  features
  You can check status of Peer with Asterisk Management Interface 
  (AMI)www.voip-info.org/wiki-Asterisk+manager+APICheers,Giovanni 
  Miano
  2005/12/30, Dov Bigio [EMAIL PROTECTED]:
  
Hi,
I am using the Queue application for 5 queues I have in my Call 
Center, and will by the end of January, implement it for the rest of the 
company (another 10 queues).

One of the main problems I face and my call center managers are worried 
about is the fact that when an agent uses the DND button of the Softphone, 
call center managers have no way of monitoring this.

Is there a way to track this?

Thank you
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-- Giovanni Miano 

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RE: [Asterisk-Users] Regular Crashes

2005-12-30 Thread Zafer Khodr









I have been experiencing a similar problem.

I have not yet been able to figure out
what the exact problem is but I know that the errors are inconsitant.

Sometimes nothing for 2 days and sometimes
5 times a day.



I thought about it a lot and I have found
only one thing in common.



The area where my server is stored gets
pretty stuffy, especially on a hot day.



I occasionally turn on the aircon as I
need to go in and do some work.

From my best recollection the server has
never crashed when the aircon has been on.

This is my third day of testing my theory,
and with the aircon controlling the room tempreture to make sure it is always
nice and cool in there I have not seen any errors for 3 days (Keeping in mind
that the day I decided to try this theory by constantly keeping the room cool
my server encountered around 4 errors in just a few hours).



So to put in short I think but cant be
sure that somehow when the room gets too hot the server goes awol and somehow
causes this error.

Dont ask me how or why all I
know is that now with controlled room temp I have not had a problem.



Good Luck













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Gough
Sent: Saturday, 31 December 2005
1:43 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Regular
Crashes









I have just setup asterisk on a debian
sarge box. I am running Asterisk
1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz)
ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions
(SIP) configured all using CounterPath(Xten) eyebeam softphone.

After many hours of Googling I have finally got it all setup and
working. We can transfer calls internally and make and receive external
calls. Its all great except for stability issues!!

Essentially every now and again, asterisk simply dies (2-3 times a
day). No warning, no error, just my console session outputs a
disconnected from console message.

Sometimes the crashes happen when you are on a call, other times when
there is no-one in the office.

The server is a brand new AMD 3400+ with 512Mb RAM. The other issue
experienced is occasional break up on inbound sound quality.

Below are traces of the last two crashes

Any Help much appreciated

Regards

Andrew Gough

FIRST TRACE

#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at lock.h:597
No locals.
#2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at channel.c:671
 f = {frametype = 4, subclass = 1,
datalen = 0, samples = 0,
 mallocd = 0, offset = 0, src = "" data = "" delivery = {tv_sec =
0,
 tv_usec = 0}, prev = 0x0, next = 0x0}
#3 0x408fc2d9 in __sip_autodestruct (data="" at chan_sip.c:1315
 p = (struct sip_pvt *) 0x81be208
#4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373
 current = (struct sched *) 0x8174868
 tv = {tv_sec = 1135275568, tv_usec =
989877}
 x = 0
 res = 1083432672
#5 0x40927e28 in do_monitor (data="" at chan_sip.c:11253
 res = 0
 sip = (struct sip_pvt *) 0x0
 peer = (struct sip_peer *) 0x0
 t = 1135275568
 fastrestart = 0
 lastpeernum = -1
 curpeernum = 6
 reloading = 0
#6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#7 0x401ac18a in clone () from /lib/tls/libc.so.6
No symbol table info available.


SECOND TRACE

#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x120010c) at lock.h:597
No locals.
#2 0x0806175a in ast_queue_hangup (chan=0x1200040) at channel.c:671
 f = {frametype = 4, subclass = 1,
datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = "">
 data = "" delivery = {tv_sec = 0, tv_usec = 0}, prev = 0x0, next =
0x0}
#3 0x408fc2d9 in __sip_autodestruct (data="" at chan_sip.c:1315
 p = (struct sip_pvt *) 0x81eb518
#4 0x08056c3e in ast_sched_runq (con=0x8172f78) at sched.c:373
 current = (struct sched *) 0x8174528
 tv = {tv_sec = 1135343875, tv_usec =
693503}
 x = 1
 res = 0
#5 0x40927e28 in do_monitor (data="" at chan_sip.c:11253
 res = 0
 sip = (struct sip_pvt *) 0x0
 peer = (struct sip_peer *) 0x0
 t = 1135343875
 fastrestart = 0
 lastpeernum = -1
 curpeernum = 6
 reloading = 0
#6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#7 0x401ac18a in clone () from /lib/tls/libc.so.6
No symbol table info available.










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Re: [Asterisk-Users] Semi-OT: porting numbers away


 Since the last hurricane (that left me without phone for around 3 weeks 
 or so), I did the call forwarding (remote call forwarding in fact).
 Lucky I was running in the cable modem in a couple of days (power restored).
 I was planning in having two DIDs in distinct providers (I've been using 
 them for outbound in BYOD contracts), and keep the POTS number in the 
 more stable one.
 But I'm still concerned with the 911 issue.
 Is the POTS telco (Bellsouth in South Florida in my case) mandated to 
 provide 911 in a ported line ? I'm not that confident in using 911 via 
 the ITSP.

When you port a number to an itsp, all calls (including 911 calls) are
handled by that itsp. There are not that many itsp's that actually handle
911 calls today, but you can certainly ask your providers (or just route
a test 911 call to them to see what happens).

Absolutely none of the itsp's offer the same service levels via the
Interent that one gets from a local telco with dedicated copper/fiber
last-mile facilities. So, expect call failures and if that's not acceptable,
keep a local pstn line or cell phone handy. For today, there are no other 
reasonable choices.


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Re: [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?

2005-12-30 Thread Moises Silva

Hello Ryan. Check out the file /etc/modules.conf, /etc/modules.d/zaptel
... if for some reason you have empty the modules.conf,
modules-update force will fix it, tough. In order to provide you with
further help, please provide more clues.

Best Regards

PD.
1. did you compiled the kernel yourself? specified modules autoload?
2. what happen when you isse modprobe wctdm after rebooting?
On 12/30/05, Ryan Booz [EMAIL PROTECTED] wrote:

Hey all…

I have a Gentoo system with Asterisk 1.2 installed. It's
been working great, for some reason the Zaptel module for my Wildcard TDM
(wctdm) seems to go missing anytime the server reboots, causing me to have to
go to the Zaptel source directory and do a quick "make install". This
is the first Linux box I've administered in a number of years Gentoo's
module stuff is a bit unfamiliar to me. Any idea what file is getting read at
boot that's taking "wctdm" out of the "modprobe"
path?

Any help on how to solve the problem would be much
appreciated. As it stands now, anytime I have to reboot the server, I have to
manually login, install the module and then start Asterisk.

Thanks!

Ryan Booz

Director of IT

Good Steward Software, LLC

111 Sowers Street, Suite 400

State College, PA 16801

Phone: 877-327-3702 x.26 (814-237-3744 x.26)

Fax: 719-623-0577

Visit us at www.energycap.com

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RE: [Asterisk-Users] PRI: This number has been disconnected

2005-12-30 Thread Javier Ergas

I restarted as you say.

PRI Debug bellow


[Kasterisk1*CLI 
-- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mdialout-trunk|1|2514990|[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m1?3:2)[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Goto (macro-dialout-trunk,s,3)
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40muser-callerid[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mDBget[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mAMPUSER=DEVICE/225/user[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- DBget: varname=AMPUSER, family=DEVICE, key=225/user
 
[Kasterisk1*CLI 
-- DBget: set variable AMPUSER to 225
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mDBget[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mAMPUSERCIDNAME=AMPUSER/225/cidname[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=225/cidname
 
[Kasterisk1*CLI 
-- DBget: set variable AMPUSERCIDNAME to sipura Linksys
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m0?5[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mSetCallerID[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40msipura Linksys 225[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mNoOp[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mUsing CallerID sipura Linksys 225[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mrecord-enable|225|OUT[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m, [1;35;40m0
 0?2:4[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Goto (macro-record-enable,s,4)
 
[Kasterisk1*CLI 
-- Executing [1;36;40mAGI[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mrecordingcheck|20051229-145347|1135878827.11[0;37;40m) in new
stack
 
[Kasterisk1*CLI 
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 
[Kasterisk1*CLI 
  recordingcheck|20051229-145347|1135878827.11: Outbound recording not
enabled
 
[Kasterisk1*CLI 
-- AGI Script recordingcheck completed, returning 0
 
[Kasterisk1*CLI 
-- Executing
[1;36;40mNoOp[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m, [1;35;40mNo
recording needed[0;37;40m) in new stack
 -- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40moutbound-callerid|1[0;37;40m) in new stack
 -- Executing
[1;36;40mDBget[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mUSEROUTCID=AMPUSER/225/outboundcid[0;37;40m) in new stack
 -- DBget: varname=USEROUTCID, family=AMPUSER, key=225/outboundcid
 -- DBget: set variable USEROUTCID to 
 -- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m1?4[0;37;40m) in new stack
 -- Goto (macro-outbound-callerid,s,4)
 -- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m1?6[0;37;40m) in new stack
 -- Goto (macro-outbound-callerid,s,6)
 -- Executing
[1;36;40mNoOp[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mCallerID set to sipura Linksys 225[0;37;40m) in new stack
 -- Executing
[1;36;40mSetGroup[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mOUT_1[0;37;40m) in new stack
 -- Executing
[1;36;40mCheckGroup[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m[0;37;40m) in new stack
 -- Executing
[1;36;40mSetVar[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mDIAL_NUMBER=2514990[0;37;40m) in new stack
 -- Executing
[1;36;40mSetVar[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mDIAL_TRUNK=1[0;37;40m) in new stack
 -- Executing
[1;36;40mAGI[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mfixlocalprefix[0;37;40m) in new stack
 
[Kasterisk1*CLI 
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
 
[Kasterisk1*CLI 
  fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf
 
[Kasterisk1*CLI 
-- AGI Script fixlocalprefix completed, returning 0
 -- Executing
[1;36;40mSetVar[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mOUTNUM=2514990[0;37;40m) in new stack
 -- Executing
[1;36;40mCut[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mcustom=OUT_1|:|1[0;37;40m) in new stack
 -- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40m0?16[0;37;40m) in new stack
 -- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mZAP/g0/2514990[0;37;40m) in new stack
 -- Making new call for cr 32772
 -- Requested transfer capability: 0x00 - SPEECH
  Protocol Discriminator: Q.931 (8)  len=49
  Call Ref: len= 2 (reference 4/0x4) (Originator)
  Message type:

Re: [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7

2005-12-30 Thread Moises Silva

hu?On 12/30/05, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi to All,Is anyone here has a settings on VM and asterisk for interconnection via SIP.ThanksLito

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[Asterisk-Users] No RTP Warning

2005-12-30 Thread William M. Sandiford




I tend to be one of 
those kind of guys that likes to eliminate all warnings. Although my 
system is running just fine, I keep getting the following 
message

Dec 30 10:39:51 
WARNING[29172]: rtp.c:779 ast_rtp_make_compatible: Channel 'IAX2/11903-16385' 
has no RTP, not doing anything

This message started 
appearing with a recent upgrade to the latest svn-trunk. Any idea as to 
what causes it and how to get rid of it?


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[Asterisk-Users] MYSQL Fetch Warning

2005-12-30 Thread William M. Sandiford





In addition to my earlier 
message about an RTP warning, I'm also getting this one a lot. My system 
is running just fine, I justkeep getting the following 
warningmessage.

Dec 30 
10:52:07 WARNING[16732]: app_addon_sql_mysql.c:316 aMYSQL_fetch: 
ast_MYSQL_fetch: numFields=7

I really don't 
understand why this message is coming up. My myql SELECT statement is 
specifically asking for 7 fields (ie I didn't do a SELECT *) and my Fetch cmd 
matches those columns exactly 1 for 1 with what is in the 
SELECT.

Any 
ideas?

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RE: [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7

2005-12-30 Thread Bill Gibbs









Voicemaster is a commercial softswitch.



www.sysmaster.com















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Friday, December 30, 2005
10:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Asterisk connect to voicemaster configuration 1.7





hu?



On 12/30/05, Angelito
Manansala [EMAIL PROTECTED]
wrote:

Hi to All,

Is anyone here has a settings on VM and asterisk for interconnection via SIP.

Thanks
Lito

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-- 
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Re: [Asterisk-Users] SNOM 360 locked up SOLVED

2005-12-30 Thread Janina Sajka

Christian Stredicke writes:
 Generally I think if people have a problem today they should move to
 4.5. This version seems to be pretty stable, we did not get any
 crash-complains or major problem reports from this version. 
 
 For those who want to move on (feature-wise), it is time to jump on the
 5.x train - the 5.0 version has been released a few days ago. We tried
 our best to test this version as good as possible (including an
 Asterisk-lab test), but from experience we know that new features always
 take a certain time to stabilise. Therefore, I would today move to 5.0
 only if it has a feature that the 4.5 does not have.

Upgraded to 5.0 earlier this week and love it. Biggest reason for the
move here was that our Snom 320 was dropping calls from time to time,
sometimes immediately on pickup, other times mid conversation. We have
not seen that since going to 5.0 earlier this week.

Janina

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[Asterisk-Users] TBCT For PRI support

2005-12-30 Thread Chris Matthews


Hi,

I'm trying to get information on what the current status of TBCT support 
in Asterisk is.


Thanks in advance,
Chris
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[Asterisk-Users] Has anyone used the applicationmap in features.conf?

2005-12-30 Thread John Voss

Has anyone used this feature?

I have been trying to find documentation on it but can't. I have tried the one 
example shown but can't get it to work.

Anyone had success?

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Re: [Asterisk-Users] Problem on ZAP channel

2005-12-30 Thread Simone Cittadini




[EMAIL PROTECTED] wrote:


Hello group members,
This is my first mail to this list. I am having one problem. When I 
dial a

number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?



First of all try to find where the delay stands.
Dial the number with the CLI open, if the delay is after the last 
pressed button and the channel coming up in the cli is a phone problem, 
look for timeouts in the configuration (on my lynksys I can force the 
sending of the number with #, dunno if it is a standard or a feature).

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[Asterisk-Users] Notifications when host fails qualify

2005-12-30 Thread Jonathan k. Creasy

I am looking to be notified via email when a host fails it's qualify (is
unreachable). I found this patch
(http://bugs.digium.com/view.php?id=5372) but I wasn't sure if I could
get that from it. 

Anyone else tried this? 

-Jonathan 
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[Asterisk-Users] using a Gigaset SX440isdn on a Diva 4BRI ?

2005-12-30 Thread Louis-David Mitterrand

Hello,

I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?

I know they probably work fine with mutliHFC cards with either bristuff
of chan_misdn but since I have some spare Divas, I was curious about
their potential as phone ports.

The Diva's 3 and 4 ports are already set to NT mode at boot time: 

/sbin/divactrl load -SeparateConfig -c 1 -f ETSI -f1 ETSI -f2 ETSI -u2 
-x2 -f3 ETSI -u3 -x3

And I think the capi.conf (using Armin's 0.6.1 version) looks OK:

[DIVA2]
ntmode=yes
isdnmode=ptp
incomingmsn=*
controller=4
group=3
accountcode=diva
context=international
echosquelch=0
echocancel=no
devices=1

But when I plug the phone into port 3 or 4 no led lights up, even with a
Y plug and when dialing I get a busy.

Before digging to deep, I am looking for some info on the feasability of
that setup.

Thanks,
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[Asterisk-Users] RPID Issue

2005-12-30 Thread Ray Van Dolson

Posted this to -dev, but it may be more appropriate here as I haven't
released my patches for it...

I've run into a couple issues relating to RPID.

I have an Asterisk 1.2.1 installation doing SIP for SPA-2002 and PAP2-NA
ATA's.  From the Asterisk box, we then do SIP to a VoIP provider who handles
the SIP to PSTN translation for us.  Pretty straight forward.

I decided to try using the RPID features in 1.2.1.  Enabled all the
trustrpid directives and sendrpid as well.  However, when I dial *81
number on my Sipura (*81 makes the call private) I get a fast busy back
from Asterisk.

Upon further investigation, it appears that Asterisk is saying the Sipura is
unauthorized.  This only happens when I try and block caller ID from the
Sipura though.

Dug around in the source a bit and it seems that Asterisk uses the contents
of the From header to authenticate the ATA against.  Normally (when making a
non-CLID blocked call), the Sipura sends a from header like the following:

From: ROY sip:[EMAIL PROTECTED];tag=cec0ff0080328e51o0

Authentication works fine in this case.

However, when the caller dials *81, the from header looks like this:

From: Anonymous sip:[EMAIL PROTECTED];tag=db61581ae353a8e1o0

I believe this is why authentication is failing.

Now, is this incorrect behavior by my ATA?  Seems like it should populate
the From header no matter what.  On the other hand, I see that the
5305715503.pw.digitalpath.net username is available in two other places in
the initial INVITE:

  * The Contact header:
Contact: Anonymous sip:[EMAIL PROTECTED]:5060
  * The RPID header:
Remote-Party-ID: ROY sip:[EMAIL 
PROTECTED];screen=yes;privacy=full;party=calling

So, what I gander is happening is that Asterisk is using the contents of the
From header the first time around to generate the auth challenge stuff
(nonce, etc) which is sent back to the ATA.  The ATA then replies with the
Proxy-Authorization field with the *correct* username (the 530571...).  This
doesn't match up with what was in the From field (Anonymous) and thus
authentication fails.  Correct?

Maybe Asterisk should initially use the username in the Contact field to do
authentication on?  Or the RPID header if available?

In any case, my solution was to modify check_user_full() and if an RPID
header is available, I copy the username out of it into the of variable and
authentication succeeds and the call works fine with or without *81.

The fix works for me, but I have a feeling there's a more correct way to
address this issue.  I'd like to know if my Sipura is misbehaving, or if
Asterisk should be looking somewhere other than the From field for
authentication info.

Thanks for any thoughts.

Ray
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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Kerry Garrison

This boils down pretty easily, how important is your phone service and how
reliable is your internet access? If you internet access is not 100% stable
and reliable then you will have problems using ITSP's. If your phone service
is critical to the operation of your business then you need to think through
the design very carefully.

Many of our clients have redundant internet connections just to keep email
running smooth, this is not the type of client I would put on an
internet-only phone system. You are only asking for trouble.

A good example is a large online retailer we just did, if they are not
getting calls, they are not getting orders, simple math there. However, 90%
of the time, they only had 3-4 lines going at the same time but during
seasonal or sale spikes this would overwhelm the 7 lines that they had. The
solution here was to use 4 analog lines and the 4th line did a call forward
on busy to Teliax pay-as-you-go which gives them 10 additional channels.
This allowed them to cut 3 phone lines off their monthly bill and double
their line capacity. Outbound calls originate on the Teliax line and fall
back to the analog lines as a backup. In the first month, they figured that
they will be saving about $600 a month in fees and long distance charges. So
their goal was accomplished, they reduced costs and increased capacity, and
they have extensions at their homes now. Sure, it's a hybrid system but this
is the beauty of Asterisk, you can choose to design a system that best fits
the needs of each individual client.

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


 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Rich Adamson
 Sent: Friday, December 30, 2005 7:34 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away
 
 
  Since the last hurricane (that left me without phone for around 3 
  weeks or so), I did the call forwarding (remote call 
 forwarding in fact).
  Lucky I was running in the cable modem in a couple of days 
 (power restored).
  I was planning in having two DIDs in distinct providers (I've been 
  using them for outbound in BYOD contracts), and keep the 
 POTS number 
  in the more stable one.
  But I'm still concerned with the 911 issue.
  Is the POTS telco (Bellsouth in South Florida in my case) 
 mandated to 
  provide 911 in a ported line ? I'm not that confident in 
 using 911 
  via the ITSP.
 
 When you port a number to an itsp, all calls (including 911 
 calls) are handled by that itsp. There are not that many 
 itsp's that actually handle
 911 calls today, but you can certainly ask your providers (or 
 just route a test 911 call to them to see what happens).
 
 Absolutely none of the itsp's offer the same service levels 
 via the Interent that one gets from a local telco with 
 dedicated copper/fiber last-mile facilities. So, expect call 
 failures and if that's not acceptable, keep a local pstn line 
 or cell phone handy. For today, there are no other reasonable choices.
 
 
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[Asterisk-Users] FOP Maximum extensions?

2005-12-30 Thread Dan Elder

I'm searching around, but not finding definative info on this, is the
maximum number of extensions available in FOP limited to 100? if so, is
there another operator console (commercial or open source) that will allow
at least 200 max extensions? The docs for FOP seem to be quite breif on
changing the layout.

thx in advance for any insight

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RE: [Asterisk-Users] FOP Maximum extensions?

2005-12-30 Thread Kerry Garrison

Its not that hard to modify the FOP settings but there is a limit to what
you can accomplish because of screen size vs readability. You can only make
buttons so small before the become unusable. 
-Kerry


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dan Elder
 Sent: Friday, December 30, 2005 9:21 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] FOP Maximum extensions?
 
 I'm searching around, but not finding definative info on 
 this, is the maximum number of extensions available in FOP 
 limited to 100? if so, is there another operator console 
 (commercial or open source) that will allow at least 200 max 
 extensions? The docs for FOP seem to be quite breif on 
 changing the layout.
 
 thx in advance for any insight
 
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RE: [Asterisk-Users] Semi-OT: porting numbers away


 Thanks, but I'm looking for information on porting numbers when the current
 provider holding the numbers goes out of business and is unreachable.  Can I
 get the numbers?  The business has had the same phone number for almost 30
 years and definitely can't lose the number due to some provider's
 instability.
 As most VoIP companies are relatively new and small, I'm a bit skittish
 about porting these numbers to an ITSN, then that company going out of
 business and not being able to get my numbers back.  How would that work?

Let's see if I can clearify this a little bit.

Local Number Portability (LPN) is a little different for telco's then it
is for cell phone providers, with far more politics involved in the telco
implementation.

On the telco side, the local telco provider (owner of the telephone number
to be moved) has to initiate the move since they own the NPA-NNX number.
They have to do something in their central office switch so that local
callers to that number are routed to some other facility as opposed to
another line in their switch. They also have to enter the change in a
database that is accessible via the nationwide SS7 network, so distant
callers to that number are routed to the appropriate central office.

The owners of the NPA-NNX number is important from the standpoint they are
supposed to be the only people with permissions to make SS7 database
entry changes for their NPA-NNX numbers.

The remote appropriate central office has to do something in their central
office switch to map that new telephone number to a line (not another
telephone number).

So in the above simple example, it takes changes to two telco switches and
one national database to complete the move. Keep in mind the owners of the
database only allow changes to that specific number-to-be-moved to come
from the owner of the NPA-NNX number. If the customer decides to discontinue
service (drop the moved telephone number), it is the responsibility of the
remote appropriate central office to notify the owner of the NPA-NNX, and
that owner initiates the local changes and database changes to bring that
number back into their home switch.

The political part of that involves questions such as should a customer
in the 123-456 central office calling a portable number such as 345-678
that is in the exact same central office be invoiced as a toll call?
Both the state public service commissions and the FCC have been involved
with those discussions, and there has not been any formal ruling that would
clearify all the questions. Therefore, many of the local exchange carriers
refuse to allow LNP as a result of these unanswered questions, and have been
getting by with it for well over a year.

Now, take the above example and change it so the portable number ends up at
a distant itsp. The exact same steps noted above have to be completed, but
in addition, the remote appropriate central office generally have to map
that moved number to another telephone number (most likely a DID number)
that is assigned to the itsp. This last step is very similar to how 800
numbers are handled by the local telephone companies now. The group that
is responsible for that mapping is generally the remote appropriate central
office.

So, now you have two steps required at the owning NPA-NNX central office,
and two steps at the receiving central office, and at least one final step
by the itsp (mapping the DID to a sip/iax customer).

Now to answer the original qustion: to move that same number to yet another
itsp requires the telco at the remote receiving central office to unmap
the local itsp DID map, and if the new mapping is to a different itsp that
is still in the same local area, remap the LNP number to the new itsp.

If the itsp is at some distant unrelated central office, the remote central
office has to unmap the original number, remove entries from their central
office switch, and update the national SS7 database to point the entry to
the next new central office. Then that next new central office has to 
create entries in their central office switch and possibly map that number
to some itsp DID number.

If all of that seems confusing, it is. But, you can begin to see who is
responsible for what, and how well the above steps work is highly dependent
on how well itsp technical folks communicate with telco technical folks in
each geographic area. Pile on top of that the requirement by most telcos
that no technical work will be performed without an accurate service order,
and you have the makings of a rather serious problem. (Try explaining all 
that to the telco folks that are responsible for originating the service 
order, and you have a bigger problem since those folks are not technical 
at all.)

Can it be done? Yes, but there is a very high probability of errors and
likely some degree of refusing to own the process through to completion.

Moving cell numbers from one provider to another is less of an issue primarily
because there are a lot less

[Asterisk-Users] IAX problem - Bug or Compatibility issue?

2005-12-30 Thread Aryanto Rachmad




Hello All,

I am looking for more thorough debug than 
the one provided by the command "iax2 debug". Could anybody point me a good 
documentation about this?

I have a issue with IAX connection. 
Sometimes it stucked.If so, I have to restart my asterisk through CLI 
command"restart now".

Comparing the debug messages of working and 
non working sequences, I have noticed that when it does not work, the following 
debug messages are missing:
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: 
(14?) Timestamp: 01581ms SCall: 00052 DCall: 16385 
[213.61.187.157:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: 
IAX Subclass: ACK Timestamp: 
01581ms SCall: 16385 DCall: 00052 
[213.61.187.157:4569] -- IAX2/sipdiscount_outbound-16385 
is making progress passing it to Zap/1-1Dec 30 17:12:31 DEBUG[12600]: 
chan_zap.c:4791 zt_indicate: Requested indication 14 on channel Zap/1-1Dec 
30 17:12:31 DEBUG[12600]: chan_zap.c:4857 zt_indicate: Received 
AST_CONTROL_PROGRESS on Zap/1-1Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 
003 Type: VOICE Subclass: 4 Timestamp: 01732ms 
SCall: 00052 DCall: 16385 [213.61.187.157:4569]Dec 30 17:12:31 
DEBUG[12569]: chan_iax2.c:6653 socket_read: Ooh, voice format changed to 
4Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: 
IAX Subclass: ACK Timestamp: 
01732ms SCall: 16385 DCall: 00052 [213.61.187.157:4569]
I have a few questions, especially about the following message:
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: 
(14?)
1. Is the number 14 in (14?), in decimal or hexadecimal?
2. If that is in decimal, why isit not translated into its 
descriptions, i.e. Call Progress, according to the IAX2 protocol document I have 
(Internet-Draft, Expires: July 5, 2005).
3. Why isthat numberquestion marked? Is it because asterisk was 
not sure?
4. If asterisk was not sure, so sometimes it decodes the message sometimes 
it could not, is there any debug to confirm this?

Or, am I looking at the wrong place? Which maybe the problem is so obvious 
and I missed that?

I am running asterisk on IBM xSeries 330 with the following detail:
CLI show versionAsterisk 1.2.1 built by root @ atvie-asterisk on a 
i686 running Linux on 2005-12-28 07:52:36 UTC# uname -aLinux 
atvie-asterisk 2.6.14-1.1653_FC4smp #1 SMP Tue Dec 13 21:46:01 EST 2005 i686 
i686 i386 GNU/Linux
Please find also below the detail of IAX debug messages.

Cheers,

Anto



MESSAGES WHEN IAX DOES NOT 
WORK

 -- Call accepted by 
213.61.187.147 (format ulaw) -- Format for call is 
ulawTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: 
IAX Subclass: ACK Timestamp: 
00057ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Tx-Frame 
Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 
4 Timestamp: 00080ms SCall: 16384 DCall: 00070 
[213.61.187.147:4569]Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3699 
find_tpeer: Created trunk peer for '213.61.187.147:4569'Dec 30 17:04:25 
DEBUG[12488]: chan_iax2.c:3725 iax2_trunk_queue: Expanded trunk 
'213.61.187.147:4569' to 6400 bytesRx-Frame Retry[ No] -- OSeqno: 002 
ISeqno: 003 Type: IAX Subclass: ACK 
Timestamp: 00080ms SCall: 00070 DCall: 16384 
[213.61.187.147:4569]
 
--- Some messages are missing hereTx-Frame Retry[000] -- OSeqno: 003 
ISeqno: 002 Type: IAX Subclass: LAGRQ 
Timestamp: 10008ms SCall: 16384 DCall: 00070 
[213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: 
IAX Subclass: LAGRQ Timestamp: 
10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame 
Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX 
Subclass: LAGRP Timestamp: 10016ms SCall: 16384 
DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 
004 Type: IAX Subclass: LAGRP Timestamp: 
10008ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame 
Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX 
Subclass: ACK Timestamp: 10008ms SCall: 16384 DCall: 
00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 
Type: IAX Subclass: ACK Timestamp: 
10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Rx-Frame 
Retry[ No] -- OSeqno: 004 ISeqno: 005 Type: IAX 
Subclass: HANGUP Timestamp: 10262ms SCall: 00070 
DCall: 16384 [213.61.187.147:4569] CAUSE 
CODE : 
0




MESSAGES AFTER ISSUING "CLI restart 
now" command

 -- Call accepted by 
213.61.187.157 (format ulaw) -- Format for call is 
ulawTx-Frame Retry[-01] --

RE: [Asterisk-Users] FOP Maximum extensions?

2005-12-30 Thread Douglas Garstang

Just a curiosity really. Anyone know how I can do this?

exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)

ie Play back the sound file after the phones receiving the page have answered? 
I know page is really simulating a one-way audio meetme conference, so I don't 
even know it it's possible.

Thanks.

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[Asterisk-Users] Playback after Page()

2005-12-30 Thread Douglas Garstang

Reposting. I forgot to change the subject. Oops.


Just a curiosity really. Anyone know how I can do this?

exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)

ie Play back the sound file after the phones receiving the page have answered? 
I know page is really simulating a one-way audio meetme conference, so I don't 
even know it it's possible.

Thanks.

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Re: [Asterisk-Users] PRI: This number has been disconnected

2005-12-30 Thread Andres




Q.931 (8)  len=13
 Call Ref: len= 2 (reference 4/0x4) (Terminator)
 Message type: DISCONNECT (69)

 [08 02 80 81]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
  Ext: 1  Cause: Unallocated (unassigned) number (1),
class = Normal Event (0) ]
 

You could also examine the PRI Hangup Cause variable and put your own 
message in via your dial plan.  In your case it is #1 - Unassigned Number.


List of causes can be found here: 
http://www.voip-info.org/wiki-Asterisk+variable+hangupcause


--
Andres
Technical Support
http://www.telesip.net


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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Ross C

So I guess I'm unclear on who 'owns' the number?  If my ITSP goes bust, and
I want to port my number to another phone service provider (ITSP or a Ma
Bell or something), does the porting process REQUIRE an acknowledgement from
the ITSP that is out of business?  Or, because I was the one who ported the
number to the ITSP in the first place, would I have enough
authority/authorization to get the number ported?  Would it do any good to
call my local telco and ask about porting numbers from providers that are
out of business?
Sounds like number porting is still a pretty gray area!

Thanks for all the feedback.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, December 30, 2005 7:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away


  Anyone have any info on porting numbers away from a VoIP provider to a
Ma
  Bell or the like?  Thanks!!
 
 I had a friend port his from Bell -VOIP -VOIP.  He had no trouble.
 
 I would use a couple providers.  So this way if one goes down there is a
backup.

In very general terms (at least in the US), telephone numbers that are
considered portable can be moved from one itsp to another. However, the
move process generally involves a request for that move on the part of
the receiving itsp and an acknowledgement on the part of the old itsp
(or original owner of the number).

That transfer process has had lots of problems of which some include:
- some itsp's don't have a clue how to do it
- some telco's refuse to acknowledge the transfer
- etc, etc.

Most of the larger telco's will handle such requests rather well.


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[Asterisk-Users] call sip:[EMAIL PROTECTED]

2005-12-30 Thread hgaillac-sip

Hello,

I wish to thank people who have called me to test my
config .
I have to test an IVR menu recorded in french so if
you call press * .

Thanks again
Harry






___ 
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs 
exceptionnels pour appeler la France et l'international.
Téléchargez sur http://fr.messenger.yahoo.com
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[Asterisk-Users] Vonage Sip Peering

2005-12-30 Thread Ben Higley

Has anyone sucessfully placed a call to a vonage user using one of the sip
peering networks.

I am trying to use sipbroker and use

exten = number,1,Dial(Sip/*472number@sipbroker.com)

i have even tried calling: number@sphone.vopr.vonage.net

I get the same return message from sipbroker as i do with
sphone.vopr.vonage.net. circuit is busy.. does this mean that sipbroker
is working? or just that vonage just isn't taking any calls in on their
server? ..???

Thanks.


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[Asterisk-Users] Manually Opening and Closing a Queue

2005-12-30 Thread Bud Bach









Does anyone have a snippet of extensions.conf to share where
they call a number to open or close a queue? Thanks.



-- Bud






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[Asterisk-Users] ENUM trees

2005-12-30 Thread Joao Pereira


Hello
I know there are 4 well known ENUM trees: e164.arpa , e164.org , 
e164.info and enum.org

Now... to which of these should I redirect my ENUM querys?
I read that e164.org is a free public ENUM root that works in a donation 
based system and is free for the public at large to use.

Shouldnt just exist one ENUM root?

Thanks
Joao Pereira
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Re: [Asterisk-Users] ENUM trees

On Fri, 2005-12-30 at 18:26 +, Joao Pereira wrote:
 Hello
 I know there are 4 well known ENUM trees: e164.arpa , e164.org , 
 e164.info and enum.org
 Now... to which of these should I redirect my ENUM querys?
 I read that e164.org is a free public ENUM root that works in a donation 
 based system and is free for the public at large to use.
 Shouldnt just exist one ENUM root?

I dont know what the status of this is but the guys over at freenum.org
were looking at parallel enum queries.  This would make queries to
multiple ENUM repositories a little faster.

You may want to see about contacting them (they are on the list as well
from what I understand so maybe they will speak up).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Playback after Page()

2005-12-30 Thread C F

I'm not sure it it's going to help for you, but try playing around
with the Local channels. and use that local channel as one of the
called devices in the page app.

On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Reposting. I forgot to change the subject. Oops.


 Just a curiosity really. Anyone know how I can do this?

 exten = page,1,SetVar(_ALERT_INFO=ring-answer)
 exten = page,2,Page(SIP/a00090101SIP/a00090301)
 exten = page,3,Playback(tt-weasels)

 ie Play back the sound file after the phones receiving the page have 
 answered? I know page is really simulating a one-way audio meetme conference, 
 so I don't even know it it's possible.

 Thanks.

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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Ross C

Thanks for the info everyone.  I think I'll just keep my numbers at my telco
and forward it to Teliax or another ITSP.  Sounds like that's the safest
thing to do.
Thx again!

-Ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Friday, December 30, 2005 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 10:19 -0600, Rich Adamson wrote:
 Let's see if I can clearify this a little bit.
 
 Local Number Portability (LPN) is a little different for telco's then it
 is for cell phone providers, with far more politics involved in the telco
 implementation.
 

And ITSPs generally are not required to let you port numbers.  Some let
you port in but not out.  Broadvoice for example has in their user
agreement that if you port a number in and they like it they can prevent
you from porting it out, if you cancel you lose the number.  Read the
agreements carefully, if its not in writing it doesnt count.

CLECs and ILECs largely are required to let you port your number (there
are some potential issues that cna prevent that but genereally that is a
true statement).

*Generally* if the ITSP is not a CLEC then they are buying from a
provider somewhere (CLEC/ILEC) even if not directly.  It is that LEC
that would be porting the number.  Even if the ITSP fails and goes away
the underlying carrier would still be able to get the phone number
ported.  However you arent the customer of that LEC so they may want
some assurances or outright refuse.  


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group

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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Andrew Kohlsmith

On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
 CLECs and ILECs largely are required to let you port your number (there
 are some potential issues that cna prevent that but genereally that is a
 true statement).

An interesting wrinkle I'm running against is that you cannot port numbers 
from a cellular carrier to a landline.  i.e. I can't port my cell # to a DID 
on my PRI.  I am not sure if this is just a line of bullshit fed to me from 
Bell Mobility (Canadian CDMA carrier) but I've not had the time to really dig 
in.  They claim that between cell carriers numbers are portable but not from 
cell to landline.

-A.
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[Asterisk-Users] Problem on ZAP channel

2005-12-30 Thread rbrahmbhatt

Hello Steve,
It's not incoming, its outgoing when I am experiencing delay.I can give
you the snippet of log if you wish.


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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 176

2005-12-30 Thread Bill Michaelson

I'm probably mistaken and unaware of a feature, but I thought the 
concept of dialing an agent does not exist.  An agent is not a channel, 
but rather, someone who associates themself with a station from which 
they service a queue.


You dial the queue with queue()


Message: 8
Date: Fri, 30 Dec 2005 20:04:38 +0530
From: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Can we dial agents from extensions.conf 
To: asterisk-users@lists.digium.com

Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-

exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup

where agent 12 is configured as :-

agent = 12,12, vivek

 

 



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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Matt Riddell

Ross C wrote:
 Thanks, but I'm looking for information on porting numbers when the current
 provider holding the numbers goes out of business and is unreachable.  Can I
 get the numbers?  The business has had the same phone number for almost 30
 years and definitely can't lose the number due to some provider's
 instability.
 As most VoIP companies are relatively new and small, I'm a bit skittish
 about porting these numbers to an ITSN, then that company going out of
 business and not being able to get my numbers back.  How would that work?

So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.

That way you can also keep the PSTN line for emergency calls (despite 911
services being offered by various ITSPs, you are relying on the Internet on
site being in top shape).

For example, I have seen more companies do something strange (or even
participate unknowingly in DDOS attacks) rendering their internet connection
useless.

While there are workarounds (maintain a good security policy, use QOS, dual
networks with router-based traffic control), it never pays to have a customer
unhappy (or dead in the case of a missed 911 call).

Typically most ITSPs rely on SLAs (Service Level Agreements) from upstream
providers which will effectively indemnify them in case of upstream failure, a
court case is not really useful in the prevention of the situation.

Is one POTS line really so much in the end?

We normally route outbound calls first via ourselves, and in the case of
network failures, fall back to the customer's PSTN/BRI line.  (BRI being quite
popular here in Italy).

This way they have unlimited outgoing lines and a set number of incoming lines
(we typically offer per channel on inbound DIDs).

If there is ever any problem with the DID, you can forward the PSTN number
back to a cellphone etc.

In fact, if I remember correctly NuFone (https://www.nufone.net/) in the USA
provides a service whereby they will try to route your number via voip and
fallback to an alternate number (ideal if available).

Furthermore, NuFone is one of the oldest (if not _the_ oldest) IAX provider
and has proven to be one of our most stable providers.

If you know what you're doing, NuFone would be my recommendation, if however
you need quite a bit of hand holding, I'd either recommend another provider,
or exhaustive use of the various Asterisk documentation resources.  :)

You can never guarantee a company is not going to go under, but when a company
provides a good service for an extended period of time, you can feel a little
safer.

-- 
Cheers,

Matt Riddell
___

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http://freevoip.gedameurope.com (Free Asterisk Voip Community)
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[Asterisk-Users] Fax Support

2005-12-30 Thread rbrahmbhatt

Can anyone guide me enabling fax support in asterisk. I tried spandsp
patch but was unsuccessful. Because patch for chan_sip.c was not proper
for asterisk's version 1.2.1. Can anyone help me adding fax support in
asterisk 1.2.1.


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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Ross C

Thanks Matt.
Are there limitations with call forwarding?  For example, with Teliax's pay
as you go plan you can have a whole bunch of simultaneous calls (we had 12
going the other day).  So say we get 10 or 12 calls on our telco number that
forwards to Teliax, is there a limit to the number of forwarded calls going
on at once?  Or does the telco hand-off the call to Teliax, then the telco
is no longer involved in that call?  I just don't want call forwarding to
defeat the purpose of going with an ITSN or limit my capabilities.

Also, do I need to have an actual physical analog line to use call
forwarding?  I have two numbers that I would like to forward, but I really
only need one POTS line that would be used by outgoing stuff (911, credit
card machines, etc).  So could I have 123-4567 forward to Teliax#987-6543
and 123-4568 forward to Teliax#987-6542, but only have one actual POTS line?
Or is this heavily dependent on the telco doing the forwarding?

Thanks!

-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Friday, December 30, 2005 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away

Ross C wrote:
 Thanks, but I'm looking for information on porting numbers when the
current
 provider holding the numbers goes out of business and is unreachable.  Can
I
 get the numbers?  The business has had the same phone number for almost 30
 years and definitely can't lose the number due to some provider's
 instability.
 As most VoIP companies are relatively new and small, I'm a bit skittish
 about porting these numbers to an ITSN, then that company going out of
 business and not being able to get my numbers back.  How would that work?

So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.

That way you can also keep the PSTN line for emergency calls (despite 911
services being offered by various ITSPs, you are relying on the Internet on
site being in top shape).

For example, I have seen more companies do something strange (or even
participate unknowingly in DDOS attacks) rendering their internet connection
useless.

While there are workarounds (maintain a good security policy, use QOS, dual
networks with router-based traffic control), it never pays to have a
customer
unhappy (or dead in the case of a missed 911 call).

Typically most ITSPs rely on SLAs (Service Level Agreements) from upstream
providers which will effectively indemnify them in case of upstream failure,
a
court case is not really useful in the prevention of the situation.

Is one POTS line really so much in the end?

We normally route outbound calls first via ourselves, and in the case of
network failures, fall back to the customer's PSTN/BRI line.  (BRI being
quite
popular here in Italy).

This way they have unlimited outgoing lines and a set number of incoming
lines
(we typically offer per channel on inbound DIDs).

If there is ever any problem with the DID, you can forward the PSTN number
back to a cellphone etc.

In fact, if I remember correctly NuFone (https://www.nufone.net/) in the USA
provides a service whereby they will try to route your number via voip and
fallback to an alternate number (ideal if available).

Furthermore, NuFone is one of the oldest (if not _the_ oldest) IAX provider
and has proven to be one of our most stable providers.

If you know what you're doing, NuFone would be my recommendation, if however
you need quite a bit of hand holding, I'd either recommend another provider,
or exhaustive use of the various Asterisk documentation resources.  :)

You can never guarantee a company is not going to go under, but when a
company
provides a good service for an extended period of time, you can feel a
little
safer.

-- 
Cheers,

Matt Riddell
___

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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Bogdan Moldovan

Depending on the forward type. You could put conditional or un-conditional
forwarding. As far as I know some telcos are placing restrictions on
conditional forwarding (and that depends on a case by case basis) but for
un-conditional forwarding I don't see why there could be a limitation.

Bogdan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Friday, December 30, 2005 9:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away

Thanks Matt.
Are there limitations with call forwarding?  For example, with Teliax's pay
as you go plan you can have a whole bunch of simultaneous calls (we had 12
going the other day).  So say we get 10 or 12 calls on our telco number that
forwards to Teliax, is there a limit to the number of forwarded calls going
on at once?  Or does the telco hand-off the call to Teliax, then the telco
is no longer involved in that call?  I just don't want call forwarding to
defeat the purpose of going with an ITSN or limit my capabilities.

Also, do I need to have an actual physical analog line to use call
forwarding?  I have two numbers that I would like to forward, but I really
only need one POTS line that would be used by outgoing stuff (911, credit
card machines, etc).  So could I have 123-4567 forward to Teliax#987-6543
and 123-4568 forward to Teliax#987-6542, but only have one actual POTS line?
Or is this heavily dependent on the telco doing the forwarding?

Thanks!

-ross

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Friday, December 30, 2005 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away

Ross C wrote:
 Thanks, but I'm looking for information on porting numbers when the
current
 provider holding the numbers goes out of business and is unreachable.  
 Can
I
 get the numbers?  The business has had the same phone number for 
 almost 30 years and definitely can't lose the number due to some 
 provider's instability.
 As most VoIP companies are relatively new and small, I'm a bit 
 skittish about porting these numbers to an ITSN, then that company 
 going out of business and not being able to get my numbers back.  How
would that work?

So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.

That way you can also keep the PSTN line for emergency calls (despite 911
services being offered by various ITSPs, you are relying on the Internet on
site being in top shape).

For example, I have seen more companies do something strange (or even
participate unknowingly in DDOS attacks) rendering their internet connection
useless.

While there are workarounds (maintain a good security policy, use QOS, dual
networks with router-based traffic control), it never pays to have a
customer unhappy (or dead in the case of a missed 911 call).

Typically most ITSPs rely on SLAs (Service Level Agreements) from upstream
providers which will effectively indemnify them in case of upstream failure,
a court case is not really useful in the prevention of the situation.

Is one POTS line really so much in the end?

We normally route outbound calls first via ourselves, and in the case of
network failures, fall back to the customer's PSTN/BRI line.  (BRI being
quite popular here in Italy).

This way they have unlimited outgoing lines and a set number of incoming
lines (we typically offer per channel on inbound DIDs).

If there is ever any problem with the DID, you can forward the PSTN number
back to a cellphone etc.

In fact, if I remember correctly NuFone (https://www.nufone.net/) in the USA
provides a service whereby they will try to route your number via voip and
fallback to an alternate number (ideal if available).

Furthermore, NuFone is one of the oldest (if not _the_ oldest) IAX provider
and has proven to be one of our most stable providers.

If you know what you're doing, NuFone would be my recommendation, if however
you need quite a bit of hand holding, I'd either recommend another provider,
or exhaustive use of the various Asterisk documentation resources.  :)

You can never guarantee a company is not going to go under, but when a
company provides a good service for an extended period of time, you can feel
a little safer.

--
Cheers,

Matt Riddell
___

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[Asterisk-Users] Which Asterisk GUI?

2005-12-30 Thread Ken D'Ambrosio

There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).

However, I'm not sure which to use.  A lot seem to be fairly
comprehensive... but until I kick the tires, it's trial-and-error.  And
that would be a *lot* of trial-and-error.

So, here's what I'm looking for:

- GPL (not a dealbreaker, but I like being able to tweak things if they
don't work the way I want)
- Comprehensive (does the substantial majority of configuration)
- Decent documentation
- Wishlist: comes with CLI tools for easy automation


I've used AMP, and found it to be reasonably decent, but there are a lot
of things it doesn't do, too.

So: which GUI do -you- like?

Thanks!

-Ken D'Ambrosio

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Re: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
 On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
  CLECs and ILECs largely are required to let you port your number (there
  are some potential issues that cna prevent that but genereally that is a
  true statement).
 
 An interesting wrinkle I'm running against is that you cannot port numbers 
 from a cellular carrier to a landline.  i.e. I can't port my cell # to a DID 
 on my PRI.  I am not sure if this is just a line of bullshit fed to me from 
 Bell Mobility (Canadian CDMA carrier) but I've not had the time to really dig 
 in.  They claim that between cell carriers numbers are portable but not from 
 cell to landline.

You can but no one is required to so most dont.  Generally speaking no
one will want to touch that becuase of potential problems.  

Its not technically impossible but it is not likely to happen for other
reasons. 


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Brian Capouch


Matt Riddell wrote:



So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.



I thought my local telco told me that if I were to do that, I would have 
to pay them LD charges for each call that came in to that number.


Or am I misunderstanding what you mean by forward here?

B.
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RE: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 13:33 -0600, Ross C wrote:
 Thanks Matt.
 Are there limitations with call forwarding?  For example, with Teliax's pay
 as you go plan you can have a whole bunch of simultaneous calls (we had 12
 going the other day).  So say we get 10 or 12 calls on our telco number that
 forwards to Teliax, is there a limit to the number of forwarded calls going
 on at once?  Or does the telco hand-off the call to Teliax, then the telco
 is no longer involved in that call?  I just don't want call forwarding to
 defeat the purpose of going with an ITSN or limit my capabilities.
 

Depends on your carrier.  I have gotten 99 forwards off an analog line
(which had no features so it was less than $10/mo).  The telco refused
to forward more than 99 concurrent calls ...  Sometimes its all in who
you ask.


 Also, do I need to have an actual physical analog line to use call
 forwarding?  I have two numbers that I would like to forward, but I really
 only need one POTS line that would be used by outgoing stuff (911, credit
 card machines, etc).  So could I have 123-4567 forward to Teliax#987-6543
 and 123-4568 forward to Teliax#987-6542, but only have one actual POTS line?
 Or is this heavily dependent on the telco doing the forwarding?

Some carriers will forward without a line others wont.  You may need to
pick and choose if  that is what you want.  

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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RE: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
 Depending on the forward type. You could put conditional or un-conditional
 forwarding. As far as I know some telcos are placing restrictions on
 conditional forwarding (and that depends on a case by case basis) but for
 un-conditional forwarding I don't see why there could be a limitation.

Well they generally like limitations because people sometimes show
questionable judgement.  A forwards to B, B forwards to A.  Call comes
in on either and you rapidly exhaust capacity.  Sometimes its just lack
of knowledge that leads people to do this sometimes they just dont think
beforehand.  

For reasons like these they like to put caps on it, but you can
generally get the caps high enough that if you are forwarding from an
analog line it shouldnt matter (ie if you need 1000 forwards you need to
reevaluate how you are doing this).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Jerry Jones



On Dec 30, 2005, at 1:48 PM, trixter aka Bret McDanel wrote:


On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:

On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number  
(there
are some potential issues that cna prevent that but genereally  
that is a

true statement).


An interesting wrinkle I'm running against is that you cannot port  
numbers
from a cellular carrier to a landline.  i.e. I can't port my cell  
# to a DID
on my PRI.  I am not sure if this is just a line of bullshit fed  
to me from
Bell Mobility (Canadian CDMA carrier) but I've not had the time to  
really dig
in.  They claim that between cell carriers numbers are portable  
but not from

cell to landline.


You can but no one is required to so most dont.  Generally speaking no
one will want to touch that becuase of potential problems.

Its not technically impossible but it is not likely to happen for  
other

reasons.


We have successfully ported cell numbers.

However from your above statement - porting points the number  
directly to you and becomes a DID.


Pointing a cell number to an existing DID would actually be a forward  
which may or may not involve a port, and does involve many other issues.



Of course I am not in Canada either.

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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Jerry Jones

Of course most carriers these days charge extra per call path. So how  
many simultaneous calls do you really need up?

On Dec 30, 2005, at 1:55 PM, trixter aka Bret McDanel wrote:


On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
Depending on the forward type. You could put conditional or un- 
conditional

forwarding. As far as I know some telcos are placing restrictions on
conditional forwarding (and that depends on a case by case basis)  
but for
un-conditional forwarding I don't see why there could be a  
limitation.


Well they generally like limitations because people sometimes show
questionable judgement.  A forwards to B, B forwards to A.  Call comes
in on either and you rapidly exhaust capacity.  Sometimes its just  
lack
of knowledge that leads people to do this sometimes they just dont  
think

beforehand.

For reasons like these they like to put caps on it, but you can
generally get the caps high enough that if you are forwarding from an
analog line it shouldnt matter (ie if you need 1000 forwards you  
need to

reevaluate how you are doing this).


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] Which Asterisk GUI?

2005-12-30 Thread Ariel Batista


Ken D'Ambrosio wrote:

There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).

However, I'm not sure which to use.  A lot seem to be fairly
comprehensive... but until I kick the tires, it's trial-and-error.
And that would be a *lot* of trial-and-error.

So, here's what I'm looking for:

- GPL (not a dealbreaker, but I like being able to tweak things if
they don't work the way I want)
- Comprehensive (does the substantial majority of configuration)
- Decent documentation
- Wishlist: comes with CLI tools for easy automation


Other then writing your own the best one I have found so far is AMP.  And 
belive me you can do allot with it. There are lots of ways to do things in 
AMP with it's custom config files.  And it's GPL and you can write your own 
changes and even add them to the project. You should look at how there 
working on version 2.0 of AMP it's going to be a major change.




I've used AMP, and found it to be reasonably decent, but there are a
lot of things it doesn't do, too.

So: which GUI do -you- like?

Thanks!

-Ken D'Ambrosio

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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread John Novack

trixter aka Bret McDanel wrote:

On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:

On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:

CLECs and ILECs largely are required to let you port your number (there are
some potential issues that cna prevent that but genereally that is a true
statement).

An interesting wrinkle I'm running against is that you cannot port numbers from a cellular carrier to a landline. i.e. I can't port my cell # to a DID on my PRI. I am not sure if this is just a line of bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but I've not had the time to really dig in. They claim that between cell carriers numbers are portable but not from
cell to landline.

You can but no one is required to so most dont. Generally speaking no one will want to touch that becuase of potential problems.

Its not technically impossible but it is not likely to happen for other reasons.

In the US that isn't the case.
LNP between wire and wireless is allowed and required.
I have ported a RCF landline number to wireless ( it took 3 months
because I put the order in on day 1 ) but it finally took.
I also ported a wireless number to Vonage, which also took 3 months, but
a year later. No excuse but it finally happened.
If I am ever able to rescue that number from Vonage is unknown, but for
now it works.

I feel sure the rules and results are different in every jurisdiction.
Is LNP even allowed in the UK or the EU?

John Novack

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RE: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Bogdan Moldovan

This is a possible scenario indeed. But this scenario should be handled by
the switches of the telco...
bogdan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Friday, December 30, 2005 9:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
 Depending on the forward type. You could put conditional or 
 un-conditional forwarding. As far as I know some telcos are placing 
 restrictions on conditional forwarding (and that depends on a case by 
 case basis) but for un-conditional forwarding I don't see why there could
be a limitation.

Well they generally like limitations because people sometimes show
questionable judgement.  A forwards to B, B forwards to A.  Call comes in on
either and you rapidly exhaust capacity.  Sometimes its just lack of
knowledge that leads people to do this sometimes they just dont think
beforehand.  

For reasons like these they like to put caps on it, but you can generally
get the caps high enough that if you are forwarding from an analog line it
shouldnt matter (ie if you need 1000 forwards you need to reevaluate how you
are doing this).

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group

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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

John Novack wrote:
 
 
 trixter aka Bret McDanel wrote:
 
 On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
  

 On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
   

 CLECs and ILECs largely are required to let you port your number
 (there are some potential issues that cna prevent that but
 genereally that is a true statement).
 

 An interesting wrinkle I'm running against is that you cannot port
 numbers from a cellular carrier to a landline.  i.e. I can't port my
 cell # to a DID on my PRI.  I am not sure if this is just a line of
 bullshit fed to me from Bell Mobility (Canadian CDMA carrier) but
 I've not had the time to really dig in.  They claim that between cell
 carriers numbers are portable but not from cell to landline.
   


 You can but no one is required to so most dont.  Generally speaking no
 one will want to touch that becuase of potential problems. 
 Its not technically impossible but it is not likely to happen for
 other reasons.  

 In the US that isn't the case.
 LNP between wire and wireless is allowed and required.
 I have ported a RCF landline number to wireless ( it took 3 months
 because I put the order in on day 1 ) but it finally took.
 I also ported a wireless number to Vonage, which also took 3 months, but
 a year later. No excuse but it finally happened.
 If I am ever able to rescue that number from Vonage is unknown, but for
 now it works.
 I feel sure the rules and results are different in  every jurisdiction.
 Is LNP even allowed in the UK or the EU?
 

Within the UK, Number Portability between providers of the same type of
service is a legal requirement.  Since we charge differently for calls
on landlines and mobiles, you cannot port mobile numbers to landlines or
 landlines to mobiles.

- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

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RE: [Asterisk-Users] Problem on ZAP channel

2005-12-30 Thread Steve Totaro

Logs are always helpful.  Are you using AMP?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Friday, December 30, 2005 2:10 PM
 To: Asterisk-Users@lists.digium.com
 Subject: [Asterisk-Users] Problem on ZAP channel
 
 Hello Steve,
 It's not incoming, its outgoing when I am experiencing delay.I can
give
 you the snippet of log if you wish.
 
 
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[Asterisk-Users] Passing authentication to an analog adapter

2005-12-30 Thread Philip Edelbrock



This is more of a curiosity and a thought than serious issue.  But, I 
wonder if I can get my Asterisk server to authenticate to my provider by 
throwing the authentication requests to the SIP analog-adapter they 
shipped me? (And I can't get in and see the authentication credentials 
in the adapter, of course.)


In other words, say I've got something like:

SIP
analog adapter -- * server -- provider

Such that the DHCP and DNS information the SIP adapter gathers comes 
from the asterisk server where it pretends to be the provider.


So then this happens:

- * tries to register with the provider
- gets the challenge
- the challenge is passed to the SIP adapter
- which answers the challenge correctly
- then * passes the answer to the provider and completes the registration

(AKA man-in-the-middle)

Is there such an authentication feature in *?  If not, I could see it 
being handy.


An alternative (although not as fun) would be to connect the analog 
adapter to a FXO card in the * server.


PS- I know some providers could treat this as a violation to their 
service agreement, btw.



Phil
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Re: [Asterisk-Users] Which Asterisk GUI?

2005-12-30 Thread George Pajari




There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).

However, I'm not sure which to use





Other then writing your own the best one I have found so far is AMP



AMP is great if the way it does things is the way you want to do it. And 
for many of our customers it has been the best way to go. The problem 
with AMP, which the authors well understand and are addressing in AMP 2, 
is that a lot of the logic and assumptions are hardwired into the config 
files it generates. Since it's Open Source you can change it if you want 
but the work can quickly escalate all out of proportion to the magnitude 
of the change. AMP 2 is supposed to be template driven which ought to 
make it easier to change the underlying implementation without the grief 
of AMP 1.


--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
 www.netvoice.ca  www.ip-centrex.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

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Re: [Asterisk-Users] Semi-OT: porting numbers away

2005-12-30 Thread Steve Kennedy

On Fri, Dec 30, 2005 at 08:22:27PM +, Ron Wellsted wrote:

 Within the UK, Number Portability between providers of the same type of
 service is a legal requirement.  Since we charge differently for calls
 on landlines and mobiles, you cannot port mobile numbers to landlines or
  landlines to mobiles.

Which is true for ITSPs too as they are bound by the Communications Act,
which covers all telecoms providers (whether traditional or new wave,
including mobile networks).


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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[Asterisk-Users] Cheap FXS/USB terminal SE-B2K, can it work with asterisk?

2005-12-30 Thread Dan Elder

I've been searching for clever ways to add a wireless phone to our asterisk
install, I could setup ATAs on each station, but I'm wondering if something
like the SE-B2K (as seen at http://www.skype-phone.net/) can be configured
to work w/asterisk  something like SJPhone. Anyone ever played with any of
these products? I've ordered the B2K, and have the SE-P1K, but I haven't
been able to find any non skype info on these devices... the B2K looks like
it'd be a great way to do this, could it work? The sales specs on various
sites that sell these say it'll do SIP, but I haven't been able to figure
out how.

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[Asterisk-Users] Motherboard choice for large opteron based asterisk server?

2005-12-30 Thread Mike Fedyk


Hi,

I am in the process of selecting an Opteron based server and am looking 
for other's experience with various motherboards and the TE411P 4-port T1.


Right now I'm looking at a Supermicro H8DA8 based 2x dual-core Opteron 
system.


http://www.antonline.com/custom_CSE822T-H8DAR-SATA-_59.htm

And a Tyan Thunder K8S Pro based dual single-core system here.

http://www.serversdirect.com/system_dept.asp?dept_id=SD-030

Comments, suggestions and experiences wanted.
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[Asterisk-Users] Aterisk 1.2.1 zaptel module not found

2005-12-30 Thread jonny hashem

Hi:
i have compiled Asterisk 1.2.1 without any problems
,But when i've tried to load the zaptel modules by
making modprobe zaptel this message shown:
FATAL: Module zaptel not found.

Regards;
jonny


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[Asterisk-Users] NOOB: Need Help Learning How to Debug PRI (U.S.)

2005-12-30 Thread Michael Collins









Help! Ive searched through the archives and Im
spinning my wheels. Im trying to get a new PRI working with
Asterisk 1.2.1. Im getting this kind of notice from the console
whenever I dial out:



 -- Executing
Dial(SIP/Mikey-3b78, Zap/g2/5551212) in new stack

Dec 30 13:09:03 NOTICE[5657]: app_dial.c:1010
dial_exec_full: Unable to create c

hannel of type 'Zap' (cause 34 - Circuit/channel congestion)

 == Everyone is busy/congested at this time (1:0/1/0)

 == Auto fallthrough, channel 'SIP/Mikey-3b78' status
is 'CONGESTION'

 -- Executing
Dial(SIP/Mikey-e5a3, Zap/g2/5595551212) in new stack

Dec 30 13:09:12 NOTICE[5663]: app_dial.c:1010
dial_exec_full: Unable to create c

hannel of type 'Zap' (cause 34 - Circuit/channel congestion)

 == Everyone is busy/congested at this time (1:0/1/0)

 == Auto fallthrough, channel 'SIP/Mikey-e5a3' status
is 'CONGESTION'



(Im using X-Pro as a SIP client)



There isnt any activity on this PRI (that Im
aware of) so I dont think its truly congested. I dont
want to call the carrier until Im in a position to gather the necessary
data to debug. Id like to debug this, but Im not sure where
to go from here. Id really like to see the raw data on the
d-channel but I dont know how to activate logging for Q.931 messaging. (I
tried to find it on the Wiki and was unsuccessful.)



I would appreciate any suggestions you might have in helping
me get this thing working. I would also appreciate any links to existing
docs that might help me learn more about PRI debugging in Asterisk.



-Michael






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Re: [Asterisk-Users] using a Gigaset SX440isdn on a Diva 4BRI ?

2005-12-30 Thread Armin Schindler

Please don't post your question on different mailinglist seperately.
I already answered that one on the isdn4linux list.

Armin

On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
 Hello,
 
 I just received a couple SX440isdn phones and was wondering if they can
 be plugged into a Diva 4BRI port in NT mode and work with
 asterisk+chan_capi?
 
 I know they probably work fine with mutliHFC cards with either bristuff
 of chan_misdn but since I have some spare Divas, I was curious about
 their potential as phone ports.
 
 The Diva's 3 and 4 ports are already set to NT mode at boot time: 
 
   /sbin/divactrl load -SeparateConfig -c 1 -f ETSI -f1 ETSI -f2 ETSI -u2 
 -x2 -f3 ETSI -u3 -x3
 
 And I think the capi.conf (using Armin's 0.6.1 version) looks OK:
 
   [DIVA2]
   ntmode=yes
   isdnmode=ptp
   incomingmsn=*
   controller=4
   group=3
   accountcode=diva
   context=international
   echosquelch=0
   echocancel=no
   devices=1
 
 But when I plug the phone into port 3 or 4 no led lights up, even with a
 Y plug and when dialing I get a busy.
 
 Before digging to deep, I am looking for some info on the feasability of
 that setup.
 
 Thanks,
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RE: [Asterisk-Users] Can we dial agents from extensions.conf

2005-12-30 Thread Alexander Lopez

 Can you tell me how agent 12 is logging in, Zap, Iax, SIP???

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Friday, December 30, 2005 9:35 AM
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] Can we dial agents from extensions.conf 
 
 Thanks a lot Mr. Alexander Lopez for your prompt attension.
 I tried the same thing but it wouldnot happen. I use it as:-
 
 exten = 12,1,Dial(Agent/12)
 exten = 12,2,Hangup
 
 where agent 12 is configured as :-
 
 agent = 12,12, vivek
 
 After the agent is logged in on extension no12 as follows 
 Callback Agent '12' logged in on 12
 
 I try to dial 12 from another sip phone and get this:-
 -- Executing Dial(SIP/62-c24e, Agent/12) in new stack
 -- outgoing agentcall, to agent '12', on 'Local/[EMAIL PROTECTED],1'
 -- Called 12
 -- Executing Dial(Local/[EMAIL PROTECTED],2, Agent/12) 
 in new stack Dec 30 14:26:54 NOTICE[13289]: app_dial.c:1011 
 dial_exec_full: Unable to create channel of type 'Agent' 
 (cause 17 - User busy)
   == Everyone is busy/congested at this time (1:1/0/0)
 -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
   == Spawn extension (default, 12, 2) exited non-zero on 
 'Local/[EMAIL PROTECTED],2'
 -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
   == Spawn extension (default, h, 1) exited non-zero on 
 'Local/[EMAIL PROTECTED],2'
   == No one is available to answer at this time (1:0/0/0)
 -- Executing Hangup(SIP/62-c24e, ) in new stack
   == Spawn extension (inoffice, 12, 2) exited non-zero on 
 'SIP/62-c24e'
 -- Executing Hangup(SIP/62-c24e, ) in new stack
   == Spawn extension (inoffice, h, 1) exited non-zero on 'SIP/62-c24e'
 
 
 I am unable to figure out why it is happening like this. They 
 are all in the same context. Also, the agent is not busy. 
 Also, I wonder why it says Unable to creat0e chanel of type 
 'Agent' cause user busy.
 Do you have any idea why is it happening so?
 I tried to tweak in but was not successful. 
 
 
 With warm regards.
 
 Vivek J. Joshi.
 
 [EMAIL PROTECTED]
 Trikon electronics Pvt. Ltd.
 
 --Optimism is a mania for saying things are well when one is in hell.
 
 
 
 Alexander Lopez wrote:
  There are options for queues.conf to not allow callers to 
 join a queue 
  if no members are logged in, also you can 'call an agent' with the 
  agent channel, (IE agent/100)
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of 
   [EMAIL PROTECTED]
   Sent: Friday, December 30, 2005 7:17 AM
   To: asterisk-users@lists.digium.com
   Subject: [Asterisk-Users] Can we dial agents from extensions.conf
   
   Hello friends,
  I wanted to ask if we can dial agents like the way we dial 
   extensions. I wanted to try this because the  users can login and 
   others can dial them. If a person has not logged in, he isnt 
   avalaible. I dont want to put people in a queue. Has anyone tried 
   this before? I was trying to do it but was unsuccessful.
   
   Please tell me if there is a tweak or a workaround for this. 
   
   
   With warm regards.
   
   Vivek J. Joshi.
   
   [EMAIL PROTECTED]
   Trikon electronics Pvt. Ltd.
   
   --Optimism is a mania for saying things are well when one 
 is in hell.
   
   `
   
 
 
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[Asterisk-Users] Recording Calls for Specific ACD Agents

2005-12-30 Thread Douglas Garstang

Is it possible to record calls for specific ACD Agents?

From looking at queues.conf and agents.conf, it appears that all calls for a 
specific queue can be record, or all calls for all agents can be recorded.

I'd like to be able to specify that calls for a _specific_ agent are recorded. 
Case in point is a new staff member that a supervisor wants to monitor. Anyone 
know if it's possible?

Thanks,
Douglas.
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Re: [Asterisk-Users] Aterisk 1.2.1 zaptel module not found

2005-12-30 Thread Moises Silva

mm and sure you have compiled the zaptel packages and make install ?On 12/30/05, jonny hashem [EMAIL PROTECTED]
 wrote:Hi:i have compiled Asterisk 1.2.1 without any problems,But when i've tried to load the zaptel modules by
making modprobe zaptel this message shown:FATAL: Module zaptel not found.Regards;jonny__Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection around
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[Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?

2005-12-30 Thread Armin Schindler

Okay, since isdn4linux.de seems to block mails from schlund.de...

On Fri, 30 Dec 2005, Armin Schindler wrote:
 On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
  Hello,
  
  I just received a couple SX440isdn phones and was wondering if they can
  be plugged into a Diva 4BRI port in NT mode and work with
  asterisk+chan_capi?
 
Yes, I don't know any reason why it shouldn't work.
 
  I know they probably work fine with mutliHFC cards with either bristuff
  of chan_misdn but since I have some spare Divas, I was curious about
  their potential as phone ports.
  
  The Diva's 3 and 4 ports are already set to NT mode at boot time: 
  
  /sbin/divactrl load -SeparateConfig -c 1 -f ETSI -f1 ETSI -f2 ETSI -u2 
  -x2 -f3 ETSI -u3 -x3
 
looks good.
  
  And I think the capi.conf (using Armin's 0.6.1 version) looks OK:
  
  [DIVA2]
  ntmode=yes
  isdnmode=ptp
  incomingmsn=*
  controller=4
  group=3
  accountcode=diva
  context=international
  echosquelch=0
  echocancel=no
  devices=1
 
isdnmode=ptp is wrong for chan_capi 0.6, use isdnmode=did
  
  But when I plug the phone into port 3 or 4 no led lights up, even with a
  Y plug and when dialing I get a busy.
  
  Before digging to deep, I am looking for some info on the feasability of
  that setup.
 
What type of cable did you use? You need to use a crossed cable with 100 Ohm 
termination.
 
Also, what version of driver/firmware do you use?
Many enhancements are done (nt-mode too) in the latest drivers from Eicon 
source RPM.

Armin
 
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RE: [Asterisk-Users] Playback after Page()

2005-12-30 Thread Alexander Lopez

You can do something like this:

exten = pagenplay,1,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])


[page]
exten = _X.,1,Set(TIMEOUT(absolute)=180)   ; Three Minutes
exten = _X.,2,SetVar(_ALERT_INFO=ring-answer)
exten = _X.,3,Dial(SIP/${EXTEN}||A(tt-weasels))


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Friday, December 30, 2005 1:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Playback after Page()
 
 I'm not sure it it's going to help for you, but try playing 
 around with the Local channels. and use that local channel as 
 one of the called devices in the page app.
 
 On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
  Reposting. I forgot to change the subject. Oops.
 
 
  Just a curiosity really. Anyone know how I can do this?
 
  exten = page,1,SetVar(_ALERT_INFO=ring-answer)
  exten = page,2,Page(SIP/a00090101SIP/a00090301)
  exten = page,3,Playback(tt-weasels)
 
  ie Play back the sound file after the phones receiving the 
 page have answered? I know page is really simulating a 
 one-way audio meetme conference, so I don't even know it it's 
 possible.
 
  Thanks.
 
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Re: [Asterisk-Users] Semi-OT: porting numbers away


  So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
  the VoIP DID, change the forwarding to another number.
  
 
 I thought my local telco told me that if I were to do that, I would have 
 to pay them LD charges for each call that came in to that number.
 
 Or am I misunderstanding what you mean by forward here?

Your pstn line will be charge the long distance charge if you forward
your local calls to an out of area number.


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[Asterisk-Users] voicemail .wav filename

2005-12-30 Thread Michiel van Baak

Hi all,

Our asterisk servers do voicemail for our customers.
We never store them on the server but mail them to the
customer. Now every .wav file is sent as MSG0.WAV
Is it possible to give the wav file a more meaningfull name
like 20051230224900.wav (year month day hour minute
second)??

Thnx.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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

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[Asterisk-Users] Outputting human readable info on a VoIP call's quality?

2005-12-30 Thread S McGowan

Hello,

Anyone know of a program that can analyse the RTP media stream and then output a
human readable graph or other file? I'd like to be able to show jitter,
difference, and if possible, echoes and other articfacts within a file of some
sort. Ethereal can show you a graph, but cannot save it as a file for
presentation to a client.

Thank you for any help you may be able to offer.

Thanks,
SKM

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RE : [Asterisk-Users] Aterisk 1.2.1 zaptel module not found

2005-12-30 Thread f6hqz-m

Title: Message



If 
yes, search if the modules are not in an any incorrect kernel branch if you have 
several :
/lib/modules/2.6.12-1-686/zaptel/zaptel.ko
May be 
it is in another branch as :
/lib/modules/2.6.12-1-386/zaptel/zaptel.ko
If 
yes, check your configuration (headers, kernel), recompile(the best 
way)or tempt to copy the modules in the correct 
branch.

Good 
Luck !
Francois BERGERET,
France.

  
  -Message d'origine-De: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] De la part de Moises 
  SilvaEnvoyé: vendredi 30 décembre 2005 
  22:33À: Asterisk Users Mailing List - Non-Commercial 
  DiscussionObjet: Re: [Asterisk-Users] Aterisk 1.2.1 zaptel 
  module not foundmm and sure you have compiled the zaptel 
  packages and make install ?
  On 12/30/05, jonny 
  hashem [EMAIL PROTECTED] 
   wrote:
  Hi:i 
have compiled Asterisk 1.2.1 without any problems,But when i've tried to 
load the zaptel modules by making modprobe zaptel this message 
shown:FATAL: Module zaptel not 
found.Regards;jonny__Do 
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Re: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 15:41 -0600, Rich Adamson wrote:
   So use call forwarding from the Telco, forward it to a VoIP DID, if you 
   lose
   the VoIP DID, change the forwarding to another number.
   
  
  I thought my local telco told me that if I were to do that, I would have 
  to pay them LD charges for each call that came in to that number.
  
  Or am I misunderstanding what you mean by forward here?
 
 Your pstn line will be charge the long distance charge if you forward
 your local calls to an out of area number.

or have business service that pays per minute.  I worked for an ISP
almost a decade ago that had many residential lines with no services and
call forwarding enabled (total cost less than $10/mo) to use to increase
their dialup numbers.  They forwarded to the main dialup number in the
hunt group.  Largely they were placed at customer sites (in exchange for
discounted service - nondialup customers).  We had 99 forwards enabled,
and becuase they were residential lines local calling meant no
additional cost.  Not a very nice thing to do, but hey after 12 years in
business that isp is still only one county large.  Kinda tells you
something about that ...


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Outputting human readable info on a VoIP call's quality?

On 12/30/05, S McGowan [EMAIL PROTECTED] wrote:
 Hello,

 Anyone know of a program that can analyse the RTP media stream and then 
 output a
 human readable graph or other file? I'd like to be able to show jitter,
 difference, and if possible, echoes and other articfacts within a file of some
 sort. Ethereal can show you a graph, but cannot save it as a file for
 presentation to a client.

 Thank you for any help you may be able to offer.


 There was a patch in the bugtracker a while back that collected rtcp
information on a call. I don't know if it made it to mainstream
Asterisk (I don't think it has yet), but that's probably a decent
start to what you're looking for. Next steps of course would be to
take that info, store it, and have a 3rd party util generate the info
you're looking for.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Recording Calls for Specific ACD Agents