date:20060324

probably a DTMF issue. Try changing it. Font have the
link here. Go to voip-info.org and search for DTMF
type. 

--- Mohammad Salaque [EMAIL PROTECTED] wrote:

 Dear List,
 I am facing another strange problem . some of my
 envisions like to use
 other prepaid card (whatever they found in market) 
 but when they dial
 that access number (phone number to put the pin)
 they get IVR (Please
 provide your pin number ) but when my user press pin
 its not going
 through, that IVR even can't get wrong pin number .
 just get
 disconnected as no pin number provided.
 
 what could be the problem ?
 
 thanks
 Salaque
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Re: [Asterisk-Users] Call terminated after 60 seconds

2006-03-24 Thread Asterisk

Nope,It's not a firewall problem.I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.It replaces the inside IP adresses from the * server in the SIP frames by the outside IP adress and creates pinholes for the udp streams.I have severalSIP connections (SIPphone, SIPGate, IPtel, Bugetphone ...) and only this one is problematic.Andre- Oorspronkelijk Bericht -Onderwerp:Re: [Asterisk-Users] Call terminated after 60 secondsAfzender: Franc
 esco
Peeters (Asterisk) [EMAIL PROTECTED]Aan:Asterisk [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comCC:[EMAIL PROTECTED] asterisk-users@lists.digium.comDatum:24-03-2006 12:18On Fri, March 24, 2006 12:01, Asterisk said: Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands)From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud ca
 ll directly to a local phone ; Inbound voicedata context ; [from-voicedata] exten = ${VOICEDATACIDNUM},1,NoOp(From Voicedata) exten = ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr) ; end of context Regards, Andre VinkCheck whether your firewall has a fixed UDP timeout set at 60 seconds...That solved my problem... ;-)(Together with activating SIP/VoIP support)-- F PeetersPIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN2 Sweex HFC-PCI cards


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Re: [Asterisk-Users] How to create [new_context] in extensions.conf?

because your phone is prob. set to a diffrent context

--- Larry Alkoff [EMAIL PROTECTED] wrote:

 Luigi Rizzo wrote:
  On Thu, Mar 23, 2006 at 01:18:15PM -0600, Larry
 Alkoff wrote:
  It _appears_ that the only way to create valid
 [context] is by a
  context = line in sip.conf.
 
  Is there another way to create a [new_context] in
 extensions.conf so I 
  can dial from it?
  
  manually with an editor ?
  or i don't understand the question.
 
 With an editor.  I've done that but extens = in the
 created [contexts] 
 are not seen or used by Asterisk so they won't dial
 out.
 
 Larry
 
 
 -- 
 Larry Alkoff N2LA - Austin TX
 Using Thunderbird on Slackware Linux
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Re: [Asterisk-Users] I'm FED UP with BroadVoice

Join the club. I just put a block on my cc. They
terminated me pretty fast :)

--- Ronald Lewis [EMAIL PROTECTED] wrote:

 After months of BroadVoice ignoring my trouble
 tickets for dropped calls,
 delayed termination, etc., I'm throwing in the
 towel. While they have
 credited $19.95 to my account, they refuse to credit
 anything more, despite
 ALL of the problems I've had. I feel the least they
 could do is credit the
 remaining $8.61 to my account, yet they won't.
 
 I haven't really been following up on porting
 between VoIP providers, but is
 there a remote chance I can save my phone number?
 I'd sure hate to change
 numbers again -- this has been a NIGHTMARE.
 Everyday, calls are dropping,
 and I'm calling people back 2 to 3 times to
 establish a decent connection.
 
 And their response (paraphrasing): We've made the
 best effort to ensure
 your service is functional ... but there are some
 things beyond our control
 with VoIP. Not good enough! I had great service
 with Vonage, and the times
 I use VoipJet, it works perfectly!
 
 Thanks in advance for any pointers.
 
 Ronald Lewis
 Denver, Colorado
 http://www.ronaldlewis.com/interviews
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[Asterisk-Users] reload - restart

2006-03-24 Thread Johann Steinwendtner


Hi !

What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting 
from new)

sip reload - (cli command sip reload). Is sip reload part of the
reload command ?

Please confirm:
Which is the correct command when adding a new SIP party in sip.conf.
sip reload  ?

Thanks !

Hans

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Re: [Asterisk-Users] I'm FED UP with BroadVoice

snip
 They have both worked reliably for me, although
 frickin' comcast 
 sometimes has very poor latency,  which they admit,
 but fail to do 
 anything about.
/snip
From what I know DSL is more reliable when it comes to
VOIP. Depending on hoy many channels you use you can
get a basic DSL line. If you have verizon they have a
plan for $15.00 a month. You get 768/128.

Dovid

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RES: [Asterisk-Users] reload - restart

2006-03-24 Thread Filipe Mordhorst

That's it. 'sip reload', reloads the sip.conf, 'extension reload' (or
extensions?), reloads the extension.conf and so on. The 'reload' command do
it all at once.

Regards,

Filipe Mordhorst  
Joinville - SC - Brasil

-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Johann
Steinwendtner
Enviada em: sexta-feira, 24 de março de 2006 08:49
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: [Asterisk-Users] reload - restart 

Hi !

What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting 
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload command ?

Please confirm:
Which is the correct command when adding a new SIP party in sip.conf.
sip reload  ?

Thanks !

Hans

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Description: S/MIME cryptographic signature
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RE: [Asterisk-Users] Voicemail limit?

2006-03-24 Thread Watkins, Bradley

I don't think there's any kind of (significantly small, anyway) limit.  I
have over 300 users at one site in voicemail.conf and no issues there.

Regards,
- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, March 22, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail limit?


Hi,
Is there an account limit for voicemail? I have 80+ users in the 
voicemail and I can only reach the 70-ieth user. If there is a limit 
how can I increase it to hundred for example?

Thanks,
Ryan

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Re: [Asterisk-Users] Asterisk Avaya Legend

2006-03-24 Thread Lacy Moore - Aspendora

Thanks Sean!

Fortunately, I don't think I will have to worry about passing extensions back and forth between Asterisk and the Legend. But, I'm glad to know that it is possible, if the situation arises. We don't have that many users. My concern is just making sure that the two can coexist for a while. We're going to use the savings from switching to the PRI to purchase the rest of the equipment, and I'll be keeping everyone connected to the Legend. Once we are able to start purchasing the IP phones, I'll move one company at a time over. During this transition, we shouldn't have to worry about extension to extension going through both.

Light bulb just went off. Looking at this:

[from-pstn]exten = 482,1,Dial(Zap/g0/482)
So if I have all 10 digits being passed and someoneplaced a call to one of our DIDs, for example, 281-604-0532, the dial plan would look like:

[from-pstn]exten = 2816040532,1,Dial(Zap/g0/2816040532)If so, I sure wish the Legend was that easy to setup! I'll probably have to get someone to set it up. The card came in yesterday for the Asterisk server, so I'll be able to start playing with things. I'm guessing I can send my own DID information to the Legend so that before the PRI even gets installed I can have everything set up and waiting.

Honestly, working with Asterisk is one of the Oh cool! moments.
On 3/23/06, Sean Cook [EMAIL PROTECTED] wrote:

-BEGIN PGP SIGNED MESSAGE-Hash: SHA1 Now, here is what I'm not sure of at this moment. For the time
being, is it possible to just pass the PRI through the Asterisk to the Legend? Will there by any type of dialplans or anything that need to be created? Will it pass the DID information through? I
was trying to look at this from the perspective that the Legend will be a channel bank, but I don't think that's a correct assumption. I think this more closely fits tying two Asterisk servers together with a T1, but I haven't been able to find any
info on this.basically you will receive the DID information and pass it straightthrough to the DS1[from-pstn]exten = 482,1,Dial(Zap/g0/482)It should be pretty straight forward... our setup is similar but the
asterisk system is configured internally not between the telco and legend. Can anyone offer any pointers, or maybe point out anything obvious that I am missing? Or, even confirm that what I'm trying to do is
possible. I have the two port card on order and would like to play with it before the PRI gets installed. I don't like working in theory on this, I'd feel much better with the equipment in hand,
even without the PRI, I can still setup a VoIP account and make sure that I can pass the call through the Asterisk into the Legend.The only thing that I am not sure of is how the legend will be passing
additional extension across the PRI so that you can dial a localextension that is on the Asterisk side...I suppose if you set up4000-4999 in UDP and pass it on the PRI you should be pretty good togo, provided that you have did's for every extension set up within the
legend.(even if they are fake did's not necessarily from the telco) Assuming all this works, I think just having Asterisk in there would solve one problem. It seems that I could set up a dial plan
so that if I dial 9 it would use the caller ID of one company, dial 8 to use the caller ID of another, etc. On the user side, they would have to dial 9 twice to place an outgoing call (since the
Legend requires a 9 for an outside line), but I think I could also set the dialplan up so that if the number dialed is a standard number, it would just use a generic caller ID number.Right but the legend also provides UDP that will allow you to specify
a range for extensions to be passed back and forth... users that dial9 and get an outside trunk will be able to match with your dialplan[from-merlin]exten = 9NXX,1,Dial(ZAP/g1/${EXTEN:-1}); where group=1 is your
telco trunkexten = 4001,1,Dial(SIP/4001) I guess I'm still confused as whether Asterisk will pass the DID information onto the Legend. If someone could point me in a direction regarding this, I'd appreciate it. I haven't found
anything, which makes me worried this isn't possible (whether because of technology or Asterisk).All you need to know is the digits that are being passed and how theyare mapped in the legend... asterisk can mimic this however you want...
-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Thunderbird -
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[Asterisk-Users] Re: How to nice agi scripts?

2006-03-24 Thread Benny Amorsen

 RS == Roger Schreiter [EMAIL PROTECTED] writes:

RS Is there any mean to let AGI scripts run in a lower priority
RS (except starting a new shell from the a short initial AGI script)?

You can start the script with renice 15 $$, or whichever value you
prefer. If the hickup happens because of the process startup, this
will not help, but at least it is easy to try out.


/Benny


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[Asterisk-Users] Cisco 7970

2006-03-24 Thread Tomislav Parčina

I have search wiki, asteriskguru, chan_sccp and some other site's for 
information's how to upgrade, and make Cisco 7970 IP phone to work with 
asterisk on SCCP firmware.

I'm sure that there are users on this group that have working Cisco 7970 phone. 
Please send me some information's how to do that.


--
Tomislav Parcina
tparcina#lama.hr
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Re: [Asterisk-Users] I'm FED UP with BroadVoice




On Mar 23, 2006, at 11:05 AM, Ronald Lewis wrote:

After months of BroadVoice ignoring my trouble tickets for dropped 
calls, delayed termination, etc., I'm throwing in the towel. While 
they have credited $19.95 to my account, they refuse to credit 
anything more, despite ALL of the problems I've had. I feel the least 
they could do is credit the remaining $8.61 to my account, yet they 
won't.
 
I haven't really been following up on porting between VoIP providers, 
but is there a remote chance I can save my phone number? I'd sure hate 
to change numbers again -- this has been a NIGHTMARE. Everyday, calls 
are dropping, and I'm calling people back 2 to 3 times to establish a 
decent connection.
 
And their response (paraphrasing): We've made the best effort to 
ensure your service is functional ... but there are some things beyond 
our control with VoIP. Not good enough! I had great service with 
Vonage, and the times I use VoipJet, it works perfectly!
 
I hate to sound like a broken record, record,  but it is also very 
important to look at the routes from you asterisk box and or phones to 
the call terminator.  No matter how good a ITSP is, or how excellent 
there support is,  if the internet between you and them has issues then 
you are SOL.


I have had positive experience with both Teliax and with Nufone.net, 
both of which I selected based on positive feedback here on the list.  
They have both worked reliably for me, although frickin' comcast 
sometimes has very poor latency,  which they admit, but fail to do 
anything about.


I am looking at also using a third company as I see that sellvoip.net is 
very inexpensive, can provide me with a local number (seattle, wa) AND 
is very close to me geographically.  The 11 MS ping from my comcast 
connection is a powerful motivator.


I too am kind of surprised Ron didn't look into teliax, particularly 
since they are just down the block (relatively speaking) from his 
location. ;)


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Re: [Asterisk-Users] Problem with MeetMe Conference!!!

On 3/24/06, serge messa [EMAIL PROTECTED] wrote:
 Hi all

 I want to use conference in Asterisk. I configure a
 conference room in meetme.conf (as conf = 600,1234)
 and extensions.conf as (exten =
 600,1,MeetMe(600,i,1234)) . When i call the extension
 600, i have the following message in the asterisk
 logs:

 WARNING[7758]: pbx.c:1688 pbx_extension_helper: No
 application 'MeetMe' for extension (conference, 600,
 1)
 == Spawn extension (conference, 600, 1) exited
 non-zero on 'IAX2/1000-2'
 -- Hungup 'IAX2/1000-2'

 I install the zaptel module with the ztdummy timer but
 the problem still exist.

 How can i do to fix this problem?


 Zaptel and ztdummy must be installed prior to building Asterisk so
that Asterisk will build and install app_meetme. If you've done this
after building Asterisk, try budiling it again now that you've
installed Zaptel and app_meetme.so should now build and install.

--
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[Asterisk-Users] Getting True ANI not Caller ID

2006-03-24 Thread Steve Totaro

I am trying get true ANI from my provider into asterisk.  I have found a few 
patches that supposedly accomplish this on Mantis and were committed to CVS 
sometime mid last year. 
 
My question is, would this have been included in stable 1.2.5 or do have to 
patch it?  There is very little info on this issue so any help or tips would be 
appreciated.
 
Thanks,
Steve Totaro
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Re: [Asterisk-Users] Asterisk Users

2006-03-24 Thread Lacy Moore - Aspendora

I'm sure the question about who uses Asterisk comes up a lot, but to me, that's something important to help the adoption of Asterisk. It's nice to see others that are using it. It always helpswhen you are presenting it to clients if you can let them know who else uses it. And, Aaron, to me, seeing as I'm from Texas, the fact that SHSU is using it means a lot, and this will mean a lot to my clients as well. I was recently in a meeting at the City of Baytown and saw that they use Cisco phones. I'm not sure if they are using Cisco Call Manager or not.

And Aaron, thanks for the info. I can understand the cost factor. What phones are you switching to?

Thanks!
Lacy

On 3/23/06, Aaron Daniel [EMAIL PROTECTED] wrote:
The 1300 phones we're moving over in the next two months are being movedoff of cisco.The reason we're moving them over is a) cost and b)
security.From the cost perspective, we're paying a yearly license feefor the servers themselves, as well as a per phone license for both theconnection to the call manager and the voicemail.Also, with the cisco
system, we've had to isolate the entire voice network since cisco onlyallows approved windows updates on their system.With Asterisk, TCO ismuch lower, the only yearly price they have to pay is the one that goes in
my pocket, and no licensing fees per phone. Win/Win if you ask me :)AaronOn Thu, 23 Mar 2006, Lacy Moore - Aspendora wrote: Just out of curiosity what was the reasons for migrating off of Cisco?This
is interesting since I've run across people who swear by Cisco, but the costs involved are just too unreasonable for me. On 3/23/06, Gary Richardson
[EMAIL PROTECTED] wrote: I'm currently 1 PRI with 25 phones. We'll be 2 PRI's and over 100 phones by mid summer. We're currently 5 DID's, but I'm pretty sure we'll be around 50 when were done.
We're currently migrating off of a cisco call manager. I recommend going slow with your transition -- do a chunk of phones, wait a week or two before doing more. Asterisk excels at interoping with other
systems; take advantage of this. Thanks. On 3/22/06, QUICK, RANDY [EMAIL PROTECTED] wrote:
Can you guys and girls give me some examples of companies using Asterisk and how many DIDs you have.I have built a small system and tested it with AASTRA 480i's and all is working perfectly.I go in front of my
Management Board tomorrow to demo the app and show them it is a viable solution.We are a medical facility with 12 facilities and a total of 1700 phones.Any
info you have would be a huge help when they ask who else is using it. Thanks in advance! Randy Quick Communications Technician II
Texoma Healthcare Systems 903.416.4398 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by
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Re: [Asterisk-Users] [OT] Polycom provisioning

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Aaron,

I have this working quite well.  Are you using FTP? or TFTP...

We are using FTP for about 40 phones and it works like a champ.   For
each phone I have...

0004f2030925.cfg
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone4710.cfg,
sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=logs/
/

then I have phone4701.cfg that contains all of the line information
and phone specific data
then the stock sip.cfg with the digitmap and global options


Sean



Aaron Daniel wrote:

 Does anyone have the polycom soundpoint ip's successfully remotely
 provisioning? I've got the phone pulling default configs, and it's
 downloading phone specific information, but it's not actually
 using that information. Any help would be appreciated :)

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Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
 
iD8DBQFEI+6ty9wPyZpnL2URAgV+AJwNrTcq6QqQAOnf+m++lteeJTaXbACeLC01
GhGc6jldP6UUcSvgwuC2GCw=
=rC2H
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Re: [Asterisk-Users] FXS channel banks

2006-03-24 Thread Chris Mason (Lists)





Tele Cost Price Reducer wrote:

  i would suggest Astribank-8 of XorCom. it is a dedicated
Asterisk compliant solution.
  you can look at : www.xorcom.com.
  
  

Looks interesting, shame they don't have a FXO version.

-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 

-- 
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dangerous content by
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Re: [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
I am currently running asterisk 1.0.9 on a system with 2 TDM400P...  I
have had fairly good success with it across the board... my only issue
is that I have monkeys who move stuff around and things get unplugged ;)

Jared Davison wrote:

 I would like to hear from anyone good or bad as what their
 experience has been in recent times with STABILITY of current
 builds of Asterisk and drivers for TDM400P.

 The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P
 cards.

 I am not concerned with: price points, or the advantages or
 disadvantages of using POTS vs ISDN technology, but simply
 RELIABILITY  stability of the Asterisk system  associated
 interface hardware and drivers.

 Do people need to reboot their systems regularly?

 Thanks in advance.


 Jared



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Re: [Asterisk-Users] Problem with MeetMe Conference!!!

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
if you have an zaptel card installed and working... try to do a load
app_meetme.so and see what happens... if it loads successfully... you
should be able to conference also check your modules.conf and make
sure you don't have noload=app_meetme.so

BJ Weschke wrote:

 On 3/24/06, serge messa [EMAIL PROTECTED] wrote:

 Hi all

 I want to use conference in Asterisk. I configure a conference
 room in meetme.conf (as conf = 600,1234) and extensions.conf as
 (exten = 600,1,MeetMe(600,i,1234)) . When i call the extension
 600, i have the following message in the asterisk logs:

 WARNING[7758]: pbx.c:1688 pbx_extension_helper: No application
 'MeetMe' for extension (conference, 600, 1) == Spawn extension
 (conference, 600, 1) exited non-zero on 'IAX2/1000-2' -- Hungup
 'IAX2/1000-2'

 I install the zaptel module with the ztdummy timer but the
 problem still exist.

 How can i do to fix this problem?


 Zaptel and ztdummy must be installed prior to building Asterisk so
 that Asterisk will build and install app_meetme. If you've done
 this after building Asterisk, try budiling it again now that you've
 installed Zaptel and app_meetme.so should now build and install.

 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/
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Re: [Asterisk-Users] Page about 70 users crash my Asterisk

On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote:
 I have here de backtrace result

 Using host libthread_db library /lib/libthread_db.so.1.
 Core was generated by `asterisk -g'.
 Program terminated with signal 11, Segmentation fault.
 #0  0xb7ece142 in ?? ()

 As I see it was in the libthread library.. So can it confirm
 my theory that is a memory problem ?

 


 There's probably far more going on there than the initial backtrace
you've got reveals. From doc/backtrace:

This document is to provide information on how to obtain the
backtraces required on the asterisk bug tracker, available at
http://bugs.digium.com. The information is required by developers to
help fix problem with bugs of any kind. Backtraces provide information
about what was wrong when a program crashed; in our case,
Asterisk. There are two kind of backtraces (aka 'bt'), which are
useful: bt and bt full.

First of all, when you start Asterisk, you MUST start it with option
-g (this tells Asterisk to produce a core file if it crashes).

If you start Asterisk with the safe_asterisk script, it automatically
starts using the option -g.

If you're not sure if Asterisk is running with the -g option, type the
following command in your shell:

debian:/tmp# ps aux | grep asterisk
root 17832  0.0  1.2   2348   788 pts/1SAug12   0:00
/bin/sh /usr/sbin/safe_asterisk
root 26686  0.0  2.8  15544  1744 pts/1SAug13   0:02
asterisk -vvvg -c
[...]

The interesting information is located in the last column.

Second, your copy of Asterisk must have been built without
optimization or the backtrace will be (nearly) unusable. This can be
done by using 'make dont-optimize' intead of 'make install' to build
and install the Asterisk binary and modules.

After Asterisk crashes, a core file will be dumped in your /tmp/
directory. To make sure it's really there, you can just type the
following command in your shell:

debian:/tmp# ls -l /tmp/core.*
-rw---  1 root root 10592256 Aug 12 19:40 /tmp/core.26252
-rw---  1 root root  9924608 Aug 12 20:12 /tmp/core.26340
-rw---  1 root root 10862592 Aug 12 20:14 /tmp/core.26374
-rw---  1 root root  9105408 Aug 12 20:19 /tmp/core.26426
-rw---  1 root root  9441280 Aug 12 20:20 /tmp/core.26462
-rw---  1 root root  8331264 Aug 13 00:32 /tmp/core.26647
debian:/tmp#

Now that we've verified the core file has been written to disk, the
final part is to extract 'bt' from the core file. Core files are
pretty big, don't be scared, it's normal.

*** NOTE: Don't attach core files on the bug tracker, we only need the
bt and bt full. ***


For extraction, we use a really nice tool, called gdb. To verify that
you have gdb installed on your system:

debian:/tmp# gdb -v
GNU gdb 6.3-debian
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and you are
welcome to change it and/or distribute copies of it under certain conditions.
Type show copying to see the conditions.
There is absolutely no warranty for GDB.  Type show warranty for details.
This GDB was configured as i386-linux.
debian:/tmp#

Which is great, we can continue. If you don't have gdb installed, go
install gdb.

Now load the core file in gdb, as follows:

debian:/tmp# gdb -se asterisk -c /tmp/core.26252
[...]
(You would see a lot of output here.)
[...]
Reading symbols from /usr/lib/asterisk/modules/app_externalivr.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app_externalivr.so
#0  0x29b45d7e in ?? ()
(gdb)

Now at the gdb prompt, type: bt
You would see output similar to:
(gdb) bt
#0  0x29b45d7e in ?? ()
#1  0x08180bf8 in ?? ()
#2  0xbcdffa58 in ?? ()
#3  0x08180bf8 in ?? ()
#4  0xbcdffa60 in ?? ()
#5  0x08180bf8 in ?? ()
#6  0x180bf894 in ?? ()
#7  0x0bf80008 in ?? ()
#8  0x180b0818 in ?? ()
#9  0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
#10 0x00a0 in ?? ()
#11 0x00a0 in ?? ()
#12 0x in ?? ()
#13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8,
filename=0x8181de8 Zap/pseudo-1324221520) at app_meetme.c:262
#14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965
#15 0xbcdffbe0 in ?? ()
#16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0
#17 0x401ec92a in clone () from /lib/libc.so.6
(gdb)


The bt's output is the information that we need on the bug tracker.

Now do a bt full as follows:
(gdb) bt full
#0  0x29b45d7e in ?? ()
No symbol table info available.
#1  0x08180bf8 in ?? ()
No symbol table info available.
#2  0xbcdffa58 in ?? ()
No symbol table info available.
#3  0x08180bf8 in ?? ()
No symbol table info available.
#4  0xbcdffa60 in ?? ()
No symbol table info available.
#5  0x08180bf8 in ?? ()
No symbol table info available.
#6  0x180bf894 in ?? ()
No symbol table info available.
#7  0x0bf80008 in ?? ()
No symbol table info available.
#8  0x180b0818 in ?? ()
No symbol table info available.
#9  0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180
No locals.
#10

Re: [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)


Jared Davison wrote:

I would like to hear from anyone good or bad as what their experience has
been in recent times with STABILITY of current builds of Asterisk and
drivers for TDM400P.

The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards.

I am not concerned with: price points, or the advantages or disadvantages of
using POTS vs ISDN technology, but simply RELIABILITY  stability of the
Asterisk system  associated interface hardware and drivers.

Do people need to reboot their systems regularly?


There are a number of folks that have reported using two TDM400's 
reliably, and a few that have indicated three working.


One of the primary issues with using either two or three cards is 
finding a motherboard that allows the two cards to use different 
interrupts (to avoid shared interrupt issues).  A second motherboard 
issue tends to be oriented around motherboards (mostly older ones now) 
that have a poor pci implementation (eg, north/south bridge chips on the 
motherboard).


Alternatives to two TDM400's include using the TDM2400 or Sangoma A200D 
where only a single pci slot is used for 1 to 24 fxo's and/or fxs's.


As of this moment, I have a single (low volume) system with both a 
TDM04b and A200D installed which will be used to compare the cards and 
provide a eval document addressing the advantages and disadvantages of 
each in certain production environments. The TDM2400 will be included in 
the mix when all the necessary cables and components are here.


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Re: [Asterisk-Users] Asterisk Avaya Legend

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Yeah... we went through the same thing... our problem was working with
asterisk it was Oh cool! then trying to make the legend work was WTF!.

Our setup is on the back side... legend still connects via PRI to the
PSTN, but we have asterisk running as a tandem off a third PRI.  Many
of the challenges are the same.  I wish we had put the money up and
gotten a 4-port T1 card and done things the way you are.  We would
have caller id and our voicemail would be much more transparent.  We
had to put a channel bank with FXO to replace the Audix voicemail
system.  While it works better than the audix system, there are some
things that I wish it did that asterisk does natively.

Sean

Lacy Moore - Aspendora wrote:

 Thanks Sean!

 Fortunately, I don't think I will have to worry about passing
 extensions back and forth between Asterisk and the Legend. But,
 I'm glad to know that it is possible, if the situation arises. We
 don't have that many users. My concern is just making sure that
 the two can coexist for a while. We're going to use the savings
 from switching to the PRI to purchase the rest of the equipment,
 and I'll be keeping everyone connected to the Legend. Once we are
 able to start purchasing the IP phones, I'll move one company at a
 time over. During this transition, we shouldn't have to worry
 about extension to extension going through both.

 Light bulb just went off. Looking at this:

 [from-pstn] exten = 482,1,Dial(Zap/g0/482)

 So if I have all 10 digits being passed and someone placed a call
 to one of our DIDs, for example, 281-604-0532, the dial plan would
 look like:

 [from-pstn] exten = 2816040532,1,Dial(Zap/g0/2816040532)

 If so, I sure wish the Legend was that easy to setup! I'll
 probably have to get someone to set it up. The card came in
 yesterday for the Asterisk server, so I'll be able to start playing
 with things. I'm guessing I can send my own DID information to the
 Legend so that before the PRI even gets installed I can have
 everything set up and waiting.

 Honestly, working with Asterisk is one of the Oh cool! moments.

 On 3/23/06, *Sean Cook* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote:



 Now, here is what I'm not sure of at this moment. For the time
 being, is it possible to just pass the PRI through the Asterisk
 to the Legend? Will there by any type of dialplans or anything
 that need to be created? Will it pass the DID information
 through? I was trying to look at this from the perspective that
 the Legend will be a channel bank, but I don't think that's a
 correct assumption. I think this more closely fits tying two
 Asterisk servers together with a T1, but I haven't been able to
 find any info on this.

 basically you will receive the DID information and pass it straight
 through to the DS1

 [from-pstn] exten = 482,1,Dial(Zap/g0/482)

 It should be pretty straight forward... our setup is similar but
 the asterisk system is configured internally not between the telco
 and legend.

 Can anyone offer any pointers, or maybe point out anything
 obvious that I am missing? Or, even confirm that what I'm trying
 to do is possible. I have the two port card on order and would
 like to play with it before the PRI gets installed. I don't like
 working in theory on this, I'd feel much better with the
 equipment in hand, even without the PRI, I can still setup a VoIP
 account and make sure that I can pass the call through the
 Asterisk into the
 Legend.

 The only thing that I am not sure of is how the legend will be
 passing additional extension across the PRI so that you can dial a
 local extension that is on the Asterisk side... I suppose if you
 set up 4000-4999 in UDP and pass it on the PRI you should be pretty
 good to go, provided that you have did's for every extension set up
 within the legend. (even if they are fake did's not necessarily
 from the telco)


 Assuming all this works, I think just having Asterisk in there
 would solve one problem. It seems that I could set up a dial plan
 so that if I dial 9 it would use the caller ID of one company,
 dial
 8 to use the caller ID of another, etc. On the user side, they
 would have to dial 9 twice to place an outgoing call (since the
 Legend requires a 9 for an outside line), but I think I could
 also set the dialplan up so that if the number dialed is a
 standard number, it would just use a generic caller ID number.

 Right but the legend also provides UDP that will allow you to
 specify a range for extensions to be passed back and forth... users
 that dial 9 and get an outside trunk will be able to match with
 your dialplan

 [from-merlin] exten = 9NXX,1,Dial(ZAP/g1/${EXTEN:-1}) ; where
 group=1 is your telco trunk exten = 4001,1,Dial(SIP/4001)




 I guess I'm still confused as whether Asterisk will pass the DID
 information onto the Legend. If someone could point me in a
 direction regarding this, I'd appreciate it. I haven't found
 anything, which makes me

[Asterisk-Users] Hints in Realtime

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Do hints work in Realtime asterisk?  not finding much on the list
archives or anywhere else for that matter... I have tried using -1
priority as mentioned once or twice but no joy

Thought?
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q+sgMrusZWUKdRWINK+ZeQI=
=6gsq
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Re: [Asterisk-Users] IAX Bridging and not recording CDR correctly

2006-03-24 Thread Matt

Hrmm.. I put that in my iax.conf... reloaded.. and I still got:

Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6 is ringing
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6
stopped sounds
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6
answered IAX2/gnutech-5
Mar 24 08:25:30 DEBUG[18185] channel.c: Avoiding initial deadlock for
'IAX2/gnutech-5'
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- Attempting native
bridge of IAX2/gnutech-5 and IAX2/calleveryone-6
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- Channel
'IAX2/gnutech-5' ready to transfer
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- Channel
'IAX2/calleveryone-6' ready to transfer
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- Releasing
IAX2/calleveryone-6 and IAX2/gnutech-5


And it recorded a call length of 1 minute.


[general]
bindport = 4569   ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
delayreject=yes
disallow=all
allow=ulaw
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1
notransfer=yes


On 3/23/06, Melcon Moraes [EMAIL PROTECTED] wrote:
 notransfer=yes

 It prevents Asterisk of getting out the media-path.

  -Original Message-
 From:   Matt [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Cc:
 Sent:  Thu, 23 Mar 2006 21:57:33 -0500
 Delivered:  Thu,  23 Mar 2006 21:12:08
 Subject:[Asterisk-Users] IAX Bridging and not recording CDR correctly

 I have a user who is off my system with IAX.  When he calls and goes
 out my long distance provider my asterisk switch seems to be bridging
 the two calls.  As a result I loose all accounting information.  All I
 get is the call setup time (15 or 20 seconds).

 How can I either make asterisk not bridge the call, or keep correct
 tabs on the call accounting for me?

 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- IAX2/calleveryone-10
 stopped sounds
 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- IAX2/calleveryone-10
 answered IAX2/gnutech-6
 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- Attempting native
 bridge of IAX2/gnutech-6 and IAX2/calleveryone-10
 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- Channel
 'IAX2/gnutech-6' ready to transfer
 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- Channel
 'IAX2/calleveryone-10' ready to transfer
 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- Releasing
 IAX2/calleveryone-10 and IAX2/gnutech-6
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 E-mail classificado pelo Identificador de Spam Inteligente Terra.
 Para alterar a categoria classificada, visite
 http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143169928.261199.12777.arrino.terra.com.br,4594,Des15,Des15


  --Original Message Ends--

 --
 Melcon Moraes [EMAIL PROTECTED]

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[Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Jordan Novak









I have compiled zaptel on Mandrake following everything I
have always done on Fedora.

It is 2.6 udev so

I had to modify the 01-devfs.rules 

Make linux26

Make

Make install

Everything appears to compile correctly but it says module
not found when doing modprobe zaptel

Is this a rights issue?



Jordan Novak








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RE: [Asterisk-Users] PSTN to Asterisk VOIP in Manila

2006-03-24 Thread Lawrence Jovellanos

Hi,

VoIP in the Philippines can be done, HOWEVER, you will be buying your E1R2
from either Globe, Bayantel, PLDT, or Digitel. You have no other choice,
most of the time you will only have 1 carrier serving the area, Philippines
has been subdivided into what is called as congressional franchise areas for
carriers.

Once these carriers detect what you are doing then may/will do things that
can make it more expensive for you if not impossible.

I have seen companies doing VoIP move from place to place  change business
names just to go around this carrier imposed hardship of doing VoIP in the
Philippines.

If the question is can it be done - YES

It is legal - now YES, before NO

Lawrence

-Original Message-
From: JP Carballo [mailto:[EMAIL PROTECTED] 
Sent: Thursday, March 23, 2006 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PSTN to Asterisk VOIP in Manila


[EMAIL PROTECTED] wrote:

 Hello,
 I'm sure you can use the Asterisk as an IP PBX.
 Good luck

 Madhawa

 Matt wrote:

 Hi list,

 Does anyone know the legalities of connecting an Asterisk box to the 
 PSTN in Manila or where I can find this info out?   I know it is 
 illegal in some countries.

 thanks

 -Matt

I posted this last year:
http://news.inq7.net/infotech/index.php?index=1story_id=57657

VoIP was declared legal last August but subject to the NTC guidelines.

Lately, I read that PLDT had lost a case against a company using grey 
routes.
Their argument was that long distance calls were their personal property 
and hence the act of not using PLDT's gateways constituted theft. It was
thrown out by the Supreme court of the Philippines.

http://www.manilastandardtoday.com/?page=news06_mar06_2006

-- 
JP Carballo

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

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

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DISCLAIMER: This email message may contain information that is confidential,
privileged, and for communication only to its intended recipient or
recipients.  If you have received this message in error, please immediately
notify the sender and delete it.

courrier electronique est confidentiel et protege. L'expediteur ne renonce
pas aux droits et obligations qui s'y rapportent. Toute diffusion,
utilisation ou copie de ce message ou des renseignements qu'il contient par
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[Asterisk-Users] getting your own phone number

2006-03-24 Thread mike webb

is there a number is the U.S that you can dial where a computer will 
reply with the phone number your calling from.

carrier is sbc if that makes any difference.
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Re: [Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Dave Cotton

On Fri, 2006-03-24 at 07:37 -0600, Jordan Novak wrote:
 I have compiled zaptel on Mandrake following everything I have always
 done on Fedora.
 
 It is 2.6 udev so…
 
 I had to modify the 01-devfs.rules 
 
 Make linux26

Isn't needed anymore if it's a recent zaptel

just 'make' is all that's needed.
 
 Make
 
 Make install…
 
 Everything appears to compile correctly but it says module not found
 when doing “modprobe zaptel”
 
 Is this a rights issue?

Only if you're not root when you do make install.


-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Doug Lytle


Jordan Novak wrote:


I have compiled zaptel on Mandrake following everything I have always 
done on Fedora.


It is 2.6 udev so…

I had to modify the 01-devfs.rules

Make linux26

Make

Make install…

Everything appears to compile correctly but it says module not found 
when doing “modprobe zaptel”


Is this a rights issue?



What version of Mandrake/Mandriva?

Doug

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RE: [Asterisk-Users] Polycom 501 and single call only using AAH 2.2

2006-03-24 Thread Jeff Herring

call waiting must be set to enabled for this extension.

I had to force the issue in AAH by doing this:

database put CW 38 ENABLED

where 38 was the extension with the issue. This forces call waiting to be
enabled respective of the AAH GUI.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Friday, March 03, 2006 4:58 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom 501 and single call only using AAH 2.2


Howdy - I've been noticing a problem where I only receive a single call, 
before other calls go to voicemail. This only happens when the user is 
on the phone. I have the polycom 501's setup for 2 lines per key and 2 
line keys for the first registration, which should allow for multiple 
calls. Anybody have any ideas what is going on?

thanks

Rolf Brusletto
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Re: [Asterisk-Users] getting your own phone number

2006-03-24 Thread Matt

SURE!
800-444-
804-883-2001
800-437-7950
866-692-6447
:)

On 3/24/06, mike webb [EMAIL PROTECTED] wrote:
 is there a number is the U.S that you can dial where a computer will
 reply with the phone number your calling from.
 carrier is sbc if that makes any difference.
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Re: [Asterisk-Users] Re: Fw: anybody has SIP realtime working ?

On 3/24/06, mustardman29 [EMAIL PROTECTED] wrote:
 So your Polycom 501's will eventually re-subscribe and BLF will eventually
 start working again after a reboot using your patch?  How long will that
 take?  Is the time to re-subscribe something you can set on the phone?

 That would be quite acceptable to me if the phone eventually re-subscribed
 on it's own without requiring a reboot.  What I am saying is that my Aastra
 9133i and Grandstream GXP2000 NEVER re-subscribe after a reboot with or
 without the patch.  I tried lot's of different settings to try make it
 happen unless I am doing something wrong or not waiting long enough for the
 phones to re-subscribe.  I must have tested it for at least 3 hours and BLF
 never came back.  I confirmed it with the Asterisk CLI as well.


 Yep. That's precisely what I'm saying. You should be able to tune the
value with minexpiry and maxexpiry settings. Be aware though that this
will also change the value/duration the phone is asked to adhere to
with regard to registrations as well. The default maxexpiry is 3600,
so if it does settle in on 3600, you must wait a full hour from the
last subscription or renewal before it tries again. You can go into
the SIP debug information on the subscribes themselves to determine
what the agreed to expiration value is.

--
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http://www.btwtech.com/
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[Asterisk-Users] pots - asterisk - tsu-600

2006-03-24 Thread mike webb

i have 6 pots lines coming in from the outside world (but we are 
reducing to 4)

all the lines have the same phone number.
i have 40 analog telephones that need to be connected to them.
one way (i think) i could do this is to have a asterisk box with a 
tdm400p with 4 fxo's connected to the pots, also in the asterisk box 
would be a pair of t100p each connected to a tsu-600, the tsu-600 would 
be filled with fxs modules giving us enough room for 48 analog lines.


another way (i think) is to have an asterisk box with just the pair of 
t100p's and one tsu-600 would have one 4 port fxo module and 5 fxs 
modules. the fxo module would be connected to the pots, the other 
tsu-600 would contain nothing but fxs modules giving us enough room for 
44 analog lines.


question is has anyone tried ether of these configurations and if so was 
it successful.

it their a better way to do it ??
please note that i realize pot lines are not the best answer but at this 
time its the way we're going to have to do it. 
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[Asterisk-Users] On ParkAndAnnounce and parking lot

2006-03-24 Thread Sharath Chandra

I am using ParkAndAnnounce to Park the call and explicitly retrieving using ParkedCall app in the dial plan. I am trying to guess the parking lot being used in a particular call by incrementing a counter just before the ParkAndAnnounce and decrement the counter just before the ParkedCall. I am not sure if this is the right way to do. What i want to know is when is the parking lot released for recycling. Is is a safe assumption to decrement just beforeParkedCall.


Thanks,
Sharath
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[Asterisk-Users] Best GUI for basic HostedPBX service

2006-03-24 Thread Michael Gaudette

Hi,

I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought
I'd pick a few brains first.

I'm not looking to configure the Asterisk server itself, VI works adequately
for that.  But I want to give Web access to as many of the following
features:

1) Voicemail configs: NIP, email to forward the .WAV file, Name of owner.  
2) Call bridging details for each extensions
a) where calls are forwarded (i.e.: SIP/account1, 555-555-1234)
b) Amount of time to wait before voicemail kicks in
c) If voicemail kicks in or not
d) Operating hours
3) A few custom variables (let's say $FOO in context [FOOBAR])


Ideally, my users would log in using their phone number, extension and
Voicemail password and be able to configure their own extension.


In your opinion, where should I start?  Is such a GUI available already, or
do I need to build my own?

Mike

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Re: [Asterisk-Users] chan_h323 problem

2006-03-24 Thread Balgansuren Batsukh

I tried many different combination of nofaststart, noh245tunneling and no 
success.


Balgaa

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

Sent: Friday, March 24, 2006 4:27 PM
Subject: Re: [Asterisk-Users] chan_h323 problem



Ganbold Tsagaankhuu wrote:

Hello,

I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.

My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN

boldsoft*CLI show version
Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686
running Linux
I can make H323 call without any problem from X-Pro and from X-lite
dead-air both end.

My default h323.conf
===
[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
allow=g723
allow=alaw
allow=ulaw
allow=gsm

gatekeeper = a.b.c.d
AllowGKRouted = yes
noFastStart = no
noH245Tunneling = no
noSilenceSuppression = no
Modified h323.conf
==
[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
allow=g723
allow=alaw
allow=ulaw
allow=gsm

gatekeeper = a.b.c.d
AllowGKRouted = yes
noFastStart = yes
noH245Tunneling = yes
noSilenceSuppression = no
I can to hear one-way audio from X-lite side, but no audio from PSTN side

I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work
default and even modified config.

Any suggestion? Which H323 channel module is better (chan_h323, oh323, 
ooh323)?


I downloaded ooh323c 0.8.1 and don't know how to create asterisk
module using source?

Regards,
Ganbold
___


Hi,

I had this same problem,
play aroud with noFastStart = yes or noFastStart = no
noH245Tunneling = yes or noH245Tunneling = no

yusuf
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[Asterisk-Users] * Meetme Freeze patch found

2006-03-24 Thread Benoit Panizzon

Hi all

Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:

http://bugs.digium.com/view.php?id=5884

Haven't tried it out yet.

Benoit Panizzon
-- 
I m p r o W a r e   A G-System Services
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__


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[Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Jordan Novak









I installed as su, and tried to compile using only make. No
problems were reported during compiling but problem persists. Any other ideas?



Jordan Novak








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[Asterisk-Users] Plain Old Answering Machine

2006-03-24 Thread Brad Glonka

Hi -

I have an fairly vanilla answering machine (actually its a combo
cordless phone/answering machine) attached to an FXS port (on a TDM400
Card).

Everything is working as planned except I seem to be having a bit of
trouble with the answering machine.  After is answers it plays the
outgoing message and then hangs up before letting the caller leave a
message. (it actually hangs up before the outgoing message is
completely played)

I think this is a problem with the answering machine (not asterisk),
but I'm limited to what I can change on the Answering Machine.

I've tried the obvious.  My dial statement does not contain a timeout...
exten = s,3,Dial,Zap/1

and I did check to make certain the machine is not in greeting only mode

There is not much too change on the answering machine other than no of rings.
I did try another another answering machine and I did not have this
problem (that's why I think its the machine not asterisk).

I think the machine is trying to detect if someone is on the remote
side of the call.  If not it hangs up.

Anything I can do with asterisk to help correct this situation?

Thanks
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RE: [Asterisk-Users] Changing codec.

2006-03-24 Thread Wai Wu




Ouch, Come on! One must know.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Wai 
WuSent: Thursday, March 23, 2006 4:25 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Changing 
codec.

Hi,

Is there a way to 
tell Asterisk to change the codec being used in the middle of an IVR 
script?
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Re: [Asterisk-Users] Asterisk Users

2006-03-24 Thread Gary Richardson

We've been running on an ICS7750 for almost 4 years. It's ridiculously
expensive. We've looked at the cost of setting up call center like
features, call recording etc and it boils down to a forklift upgrade
that will end up costing anywhere from $100-$250K. That's partially
our problem, since we bought the ICS unit in the first place.

Plus, the ICS7750 in particular has been very unreliable for us. The
switching backplain died 3 times in it. The memory for the SPE blades
was underspeced for unity (especially after SQL server memory leaks).

We never upgraded off of callmanager 3.2. We got our system barely
stable, but never working properly. It seems like every time I walk
past it in the lan room, somethings goes wrong. TAC couldn't really
solve our problems. The software itself is far too complex and
unstable. It takes 15-25 screens to configure a new user with
voicemail. We've never be able to properly use conference bridges --
they die after 3 minutes.

Possibly CCM4/5 is better. We'll never know at this point.

I suppose I'm just whining now. Cisco has tried really hard to help us out.

-- A jaded Cisco user.

On 3/23/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
 Just out of curiosity what was the reasons for migrating off of Cisco?  This
 is interesting since I've run across people who swear by Cisco, but the
 costs involved are just too unreasonable for me.



 On 3/23/06, Gary Richardson [EMAIL PROTECTED] wrote:
  I'm currently 1 PRI with 25 phones. We'll be 2 PRI's and over 100
  phones by mid summer. We're currently 5 DID's, but I'm pretty sure
  we'll be around 50 when were done.
 
  We're currently migrating off of a cisco call manager. I recommend
  going slow with your transition -- do a chunk of phones, wait a week
  or two before doing more. Asterisk excels at interoping with other
  systems; take advantage of this.
 
  Thanks.
 
  On 3/22/06, QUICK, RANDY [EMAIL PROTECTED] wrote:
  
  
   Can you guys and girls give me some examples of companies using Asterisk
 and
   how many DIDs you have.  I have built a small system and tested it with
   AASTRA 480i's and all is working perfectly.  I go in front of my
 Management
   Board tomorrow to demo the app and show them it is a viable solution.
 We
   are a medical facility with 12 facilities and a total of 1700 phones.
 Any
   info you have would be a huge help when they ask who else is using it.
  
   Thanks in advance!
  
  
   Randy Quick
   Communications Technician II
   Texoma Healthcare Systems
   903.416.4398
   [EMAIL PROTECTED]
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Re: [Asterisk-Users] Call Monitoring?

2006-03-24 Thread Gary Richardson

You could use contexts for this. By default put everyone into the
'internal' context. Managers would go into the 'managers' context,
which would include the 'internal' context.

The manager context specifically would have the exten's to monitor or
barge into calls. By including the internal context, they'd have the
same dialplan otherwise.

You determine which context a user gets by default in sip.conf (if
you're using sip phones..).

On 3/23/06, Charles Marcus [EMAIL PROTECTED] wrote:
 1. Is Asterisk capable of allowing for setting up Groups so that only
 one extension in a Group can selectively monitor one of the other
 extensions in the Group (but none of the others can initiate it)?

 This would be for Managers to listen to Sales Calls of other members of
 their Team, to provide feedback to the Rep for training purposes.

 2. Alternatively, can a Group be defined that will allow multiple
 extensions to listen in on another call in progress?

 Again, we want to use this kind of functionality to do some Sales
 Technique Training calls.

 --

 Best regards,

 Charles
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[Asterisk-Users] Re: Which Mac OSX softphone with IAX2 support?

2006-03-24 Thread Aldo Bergamini

[EMAIL PROTECTED] is believed to have said: 


Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?

thanks
Mike



Hi Mike,

look for LoudHush on VersionTracker...

HTH
Aldo

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Re: [Asterisk-Users] Dialling Problem

2006-03-24 Thread Mohammad Salaque

thanks Dovid ,

i solved that  yes it was DTMF issue .

thanks
Salaque

On 3/24/06, Dovid Bender [EMAIL PROTECTED] wrote:
 probably a DTMF issue. Try changing it. Font have the
 link here. Go to voip-info.org and search for DTMF
 type.

 --- Mohammad Salaque [EMAIL PROTECTED] wrote:

  Dear List,
  I am facing another strange problem . some of my
  envisions like to use
  other prepaid card (whatever they found in market)
  but when they dial
  that access number (phone number to put the pin)
  they get IVR (Please
  provide your pin number ) but when my user press pin
  its not going
  through, that IVR even can't get wrong pin number .
  just get
  disconnected as no pin number provided.
 
  what could be the problem ?
 
  thanks
  Salaque
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Re: [Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Andrew Latham

check your /lib/modules for a custom kernel, copy it over to your
current kernel..

On 3/24/06, Jordan Novak [EMAIL PROTECTED] wrote:



 I installed as su, and tried to compile using only make. No problems were
 reported during compiling but problem persists. Any other ideas?



 Jordan Novak


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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [Asterisk-Users] Mandrake zaptel module not found after compiling

2006-03-24 Thread Frederic Jean


Hello,

I use Mandrake 10.1 and I had no problem, just
had to install the kernel sources and follow the
instructions in README.udev

I do make linux26.

you have to reboot after you follow the instructions in REAME.udev
so it can take effect. Make sure zapata.conf is in /etc and check
for interrupt problems as well.

Frederic

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

Sent: Friday, March 24, 2006 13:35
Subject: Re: [Asterisk-Users] Mandrake zaptel module not found after 
compiling




check your /lib/modules for a custom kernel, copy it over to your
current kernel..

On 3/24/06, Jordan Novak [EMAIL PROTECTED] wrote:




I installed as su, and tried to compile using only make. No problems were
reported during compiling but problem persists. Any other ideas?



Jordan Novak


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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---








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RE: [Asterisk-Users] Best GUI for basic HostedPBX service

2006-03-24 Thread Justin Hamade

You will probably have to build that yourself, or really customize
something off the shelf.  Depending on what phones you are using you
might be able to do that via the phones xml interface.

Have fun with that I would be interested to see how it goes.

--
Justin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Gaudette
Sent: Friday, March 24, 2006 6:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Best GUI for basic HostedPBX service

Hi,

I'm looking for a web GUI to offer my end-users (Hosted PBX), and I
thought
I'd pick a few brains first.

I'm not looking to configure the Asterisk server itself, VI works
adequately
for that.  But I want to give Web access to as many of the following
features:

1) Voicemail configs: NIP, email to forward the .WAV file, Name of
owner.  
2) Call bridging details for each extensions
a) where calls are forwarded (i.e.: SIP/account1,
555-555-1234)
b) Amount of time to wait before voicemail kicks in
c) If voicemail kicks in or not
d) Operating hours
3) A few custom variables (let's say $FOO in context [FOOBAR])


Ideally, my users would log in using their phone number, extension and
Voicemail password and be able to configure their own extension.


In your opinion, where should I start?  Is such a GUI available already,
or
do I need to build my own?

Mike

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Re: [Asterisk-Users] [OT] Polycom provisioning


Yeah, that's kinda what I've got set up in mine:

APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=44198/phone.cfg, 
sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=44198 
OVERRIDES_DIRECTORY=44198 CONTACTS_DIRECTORY=44198/


It's pulling 44198/phone.cfg from the server fine, but for some reason 
it's not using the information in that file.  Can I see an example of 
someone's phone specific configuration?


Aaron

On Fri, 24 Mar 2006, Sean Cook wrote:


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Aaron,

I have this working quite well.  Are you using FTP? or TFTP...

We are using FTP for about 40 phones and it works like a champ.   For
each phone I have...

0004f2030925.cfg
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone4710.cfg,
sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=logs/
/

then I have phone4701.cfg that contains all of the line information
and phone specific data
then the stock sip.cfg with the digitmap and global options


Sean



Aaron Daniel wrote:


Does anyone have the polycom soundpoint ip's successfully remotely
provisioning? I've got the phone pulling default configs, and it's
downloading phone specific information, but it's not actually
using that information. Any help would be appreciated :)


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Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFEI+6ty9wPyZpnL2URAgV+AJwNrTcq6QqQAOnf+m++lteeJTaXbACeLC01
GhGc6jldP6UUcSvgwuC2GCw=
=rC2H
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Echo and static when dialing Asterisk

2006-03-24 Thread jglucky


I am using Asterisk 1.2.5 and when ever I dial to either another extension
or to an outside number, I seem to be experiencing a really bad echo
problem.  The echo is so bad, that Asterisk is almost unusable.
I am using VoIPJet as my outgoing IAX provider and do not use any Zaptel
hardware.

Anyone have any idea's how to suppress this echo?

Thank you,

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

DID YOU KNOW?
BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2
(Document Management) Application Worldwide.

BlueWare Market Share for Hospital Document Management Systems is in 25
states in the US.

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Re: [Asterisk-Users] Asterisk Users

I do know what you're saying about the need to know who all uses Asterisk 
is important.  It may not be a bad idea to get a definitive list together 
at some point in the future so that anyone that's trying to get approval 
can sit down and show a list of successful deployments.  Guess in time 
this may happen :)


I can't really say for sure what the City of Baytown uses.  I know the 
city of Hunstville uses CCM for their phones, which is kinda amusing since 
most of their buildings are connected over wireless, making phone calls a 
pain.


On campus, we already have about 1300 Cisco's deployed on a CCM, so those 
are going to be a quick shot over to Asterisk since provisioning from one 
to the other is fairly simple.  The rest of the phones, I think we decided 
we're going to drop Cisco 7940's into dorms since they're simple, they 
don't need much, and we know they work for what we need them to do.  Right 
now I'm in the midst of research phones for the rest of campus, Polycom 
has impressed me so far, so I'm hoping to be able to push that for campus.


Aaron

On Fri, 24 Mar 2006, Lacy Moore - Aspendora wrote:


I'm sure the question about who uses Asterisk comes up a lot, but to me,
that's something important to help the adoption of Asterisk.  It's nice to
see others that are using it.  It always helps when you are presenting it to
clients if you can let them know who else uses it.  And, Aaron, to me,
seeing as I'm from Texas, the fact that SHSU is using it means a lot, and
this will mean a lot to my clients as well.  I was recently in a meeting at
the City of Baytown and saw that they use Cisco phones.  I'm not sure if
they are using Cisco Call Manager or not.

And Aaron, thanks for the info.  I can understand the cost factor.  What
phones are you switching to?

Thanks!
Lacy



On 3/23/06, Aaron Daniel [EMAIL PROTECTED] wrote:


The 1300 phones we're moving over in the next two months are being moved
off of cisco.  The reason we're moving them over is a) cost and b)
security.  From the cost perspective, we're paying a yearly license fee
for the servers themselves, as well as a per phone license for both the
connection to the call manager and the voicemail.  Also, with the cisco
system, we've had to isolate the entire voice network since cisco only
allows approved windows updates on their system.  With Asterisk, TCO is
much lower, the only yearly price they have to pay is the one that goes in
my pocket, and no licensing fees per phone. Win/Win if you ask me :)

Aaron

On Thu, 23 Mar 2006, Lacy Moore - Aspendora wrote:


Just out of curiosity what was the reasons for migrating off of

Cisco?  This

is interesting since I've run across people who swear by Cisco, but the
costs involved are just too unreasonable for me.

On 3/23/06, Gary Richardson [EMAIL PROTECTED] wrote:


I'm currently 1 PRI with 25 phones. We'll be 2 PRI's and over 100
phones by mid summer. We're currently 5 DID's, but I'm pretty sure
we'll be around 50 when were done.

We're currently migrating off of a cisco call manager. I recommend
going slow with your transition -- do a chunk of phones, wait a week
or two before doing more. Asterisk excels at interoping with other
systems; take advantage of this.

Thanks.

On 3/22/06, QUICK, RANDY [EMAIL PROTECTED] wrote:



Can you guys and girls give me some examples of companies using

Asterisk

and

how many DIDs you have.  I have built a small system and tested it

with

AASTRA 480i's and all is working perfectly.  I go in front of my

Management

Board tomorrow to demo the app and show them it is a viable

solution.  We

are a medical facility with 12 facilities and a total of 1700

phones.  Any

info you have would be a huge help when they ask who else is using it.

Thanks in advance!


Randy Quick
Communications Technician II
Texoma Healthcare Systems
903.416.4398
[EMAIL PROTECTED]
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Call transfer - (Call failed)

2006-03-24 Thread Giuseppe


Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.

This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)

When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call and hears silence forever.
Does anyone know why?

Giuseppe
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Re: [Asterisk-Users] Cisco 7970

2006-03-24 Thread jason justman


Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/

1.) setup your /etc/asterisk/sccp.conf with something like:

[devices]

type= 7970  ; device type (see below)
autologin   = 30,31,  ; lines list. You can add an empty line for an 
empty button (7960, 7970, 7940, 7920)
description = jj7970; internal description. Not 
important

tzoffset  = -9
transfer = on   ; enable or disable the transfer 
capability. It does remove the transfer softkey
park = on   ; take a look to the 
compile howto. Park stuff is not compiled by default
speeddial = ; you can add an empty speedial 
if you want an empty button (7960, 7970, 7920)

speeddial = *97,voicemail,

cfwdall = off   ; activate the callforward stuff 
and softkeys

cfwdbusy = off
dtmfmode = inband   ; inband or outofband. 
outofband is the native cisco dtmf tone play.
   ; Some phone model does 
not play dtmf tones while connected (bug?), so the default is inband
imageversion = P00405000700 ; useful to upgrade old 
firmwares (the ones that do not load *.xml from the tftp server)

deny=0.0.0.0/0.0.0.0; Same as general
permit=192.168.1.90/255.255.255.255 ; This device can register only 
using this ip address
dnd = on; turn on the dnd 
softkey for this device. Valid values are off, on (busy signal), 
reject (busy signal), silent (ringer = silent)
trustphoneip = no   ; The phone has a ip 
address. It could be private, so if the phone is behind NAT
   ; we don't have to trust 
the phone ip address, but the ip address of the connection
;earlyrtp = none; valid options: none, 
offhook, dial, ringout. default is none.
   ; The audio strem will 
be open in the progress and connected state.
private = on; permit the private function 
softkey for this device
mwilamp = on; Set the MWI lamp style when 
MWI active to on, off, wink, flash or blink

mwioncall = off ; Set the MWI on call.
device = SEP00131A1F6366   ; device name SEPMAC

[lines]

id  = 30; future use
pin = 1234  ; future use
label   = 30; button line label (7960, 7970, 
7940, 7920)

description = Line 30   ; top diplay description
context = from-internal ; sccp
incominglimit = 2   ; more than 1 incoming 
call = call waiting.
transfer = on   ; per line transfer capability. 
on, off, 1, 0
mailbox = 30; voicemail.conf (syntax: 
[EMAIL PROTECTED]:folder])
vmnum = *97 ; speeddial for 
voicemail administration, just a number to dial

cid_name = JJJ  ; caller id name
cid_num = 30
trnsfvm = 1000  ; extension to redirect the 
caller (e.g for voicemail)
secondary_dialtone_digits = 9   ; digits for the secondary 
dialtone (max 9 digits)

secondary_dialtone_tone = 0x21  ; outside dialtone
musicclass=default  ; Sets the default music on hold 
class

language=en ; Default language setting
;accountcode=79501  ; accountcode to ease billing
rtptos = 184; sets the the rtp packets TOS 
for this line
echocancel = on ; sets the phone echocancel for 
this line
silencesuppression = off; sets the silence suppression 
for this line
;callgroup=1,3-4; We are in caller 
groups 1,3,4. Valid for this line
;pickupgroup=1,3-5  ; We can do call pick-p for call 
group 1,3,4,5. Valid for this line
;amaflags = ; Sets the default AMA flag code 
stored in the CDR record for this line

line = 30

(do the same for line 31)

2.)  setup lines 30/31 as a custom extension in astersik (i used amp) 
and had it dial SCCP/30 and SCCP/31 as needed



3.)  setup /tftpboot config for SEPMAC.xml

device  xsi:type=axl:XIPPhone
devicePool
nameDefault/name
dateTimeSetting
nameCMLocal/name
dateTemplatey-M-D/dateTemplate
timeZoneW. Europe Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member  priority=0
callManager
ports

ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName(ASTERISK IP HERE)/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo
nameEnable/name
srstOptionEnable/srstOption
userModifiabletrue/userModifiable
ipAddr1(ASTERISK IP HERE)/ipAddr1

port12000/port1
ipAddr2/ipAddr2

Re: [Asterisk-Users] RE: MeetMe freezes machine with Junghanns

2006-03-24 Thread Henning Holtschneider

On Thursday 23 March 2006 22:14, BJ Weschke wrote:

  There's been two very recent commits (one less than an hour ago) that
 may very well correct your issues.

The patch at http://bugs.digium.com/view.php?id=5884 fixes the problem!

Cheers,
Henning Holtschneider
--
LocaNet oHG - http://www.loca.net
Lindemannstrasse 81, D-44137 Dortmund
tel +49 231 91596-25, fax +49 231 91596-55


pgpoDln7bEb6t.pgp
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[Asterisk-Users] Snom 360 problems

2006-03-24 Thread Brian Kennedy


Anyone have a Snom they're happy with?   How did you manage that?  :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar 
~15 Sipura/Linsys SPA-841

~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen 
between any two.  Also everything, except the snom, is working sweetly.


The main problem is ECHO.. awful echo and only on the Snom.  When using 
a Zap line or to another sip phone.  I've tweaked the * for echo and 
managed to only create echo and piss everyone else off, pounded the 
settings in the Snom trying to find anything, and updated the firmware 
to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 
v3.36 after noticing a changelog that sounded like it may have related 
to echo.  Not even a slight reduction in echo so far.


A second serious problem is Call join.   Even with Call join on Xfer (2 
calls) OFF if the user is doing a transfer of one call when a second 
starts ringing the 2 callers get bridged, no transfer.  Really nice, now 
I have two customers talking to each other with no clue what's going on 
and neither gets who they were trying to reach.


Any ideas on what I can try next?

Thanks...
...Brian
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Re: [Asterisk-Users] Plain Old Answering Machine




I have an fairly vanilla answering machine (actually its a combo
cordless phone/answering machine) attached to an FXS port (on a TDM400
Card).

Everything is working as planned except I seem to be having a bit of
trouble with the answering machine.  After is answers it plays the
outgoing message and then hangs up before letting the caller leave a
message. (it actually hangs up before the outgoing message is
completely played)

I think this is a problem with the answering machine (not asterisk),
but I'm limited to what I can change on the Answering Machine.

I've tried the obvious.  My dial statement does not contain a timeout...
exten = s,3,Dial,Zap/1

and I did check to make certain the machine is not in greeting only mode

There is not much too change on the answering machine other than no of rings.
I did try another another answering machine and I did not have this
problem (that's why I think its the machine not asterisk).

I think the machine is trying to detect if someone is on the remote
side of the call.  If not it hangs up.

Anything I can do with asterisk to help correct this situation?


Pure guess the answering machine may be opening tip/ring and 
asterisk interprets that as a disconnect.


Try putting a plain old voltmeter across tip/ring and see what happens 
when the disconnect occurs.


Another guess is the answering machine is spewing a tone between the 
message and recording, and asterisk hears the tone and disconnects.


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FW: [Asterisk-Users] Asterisk Users

2006-03-24 Thread Bob McDowell


Perhaps a page on the wiki would work?  We could set the ground rules
similar to other industries:  no names, nothing more defining than a
region, the number of units, etc.  Would that be useful?

For example, I can describe this organization as a security company in
Southwest Missouri using asterisk with 60 sets and 16 lines.

When you strip off my name and email, it gets a little less certain who
I am talking about...


Bob McDowell



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[Asterisk-Users] Maximum Queue Name Length

2006-03-24 Thread Douglas Garstang

Do queue names have a max length? I have a queue named 'oneeighty_techsupp' in 
queues.conf, and a 'show queues' returns a truncated queue name. Is that just a 
display bug, or do queues names have a max length of 12?

demeter*CLI show queues
oneeighty_te has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), 
W:0, C:0, A:0, SL:0.0% within 0s
   Members:  
  Agent/80014255 (Not in use) has taken no calls yet
  Agent/80014257 (Not in use) has taken no calls yet

I thought the max length of a context name was 16...?

Doug.
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Re: FW: [Asterisk-Users] Asterisk Users


Perhaps a page on the wiki would work?  We could set the ground rules
similar to other industries:  no names, nothing more defining than a
region, the number of units, etc.  Would that be useful?

For example, I can describe this organization as a security company in
Southwest Missouri using asterisk with 60 sets and 16 lines.

When you strip off my name and email, it gets a little less certain who
I am talking about...


Bob McDowell


I like the idea of having the information on the wiki, makes it simpler 
for everyone to see just how well the project is doing.  I'm not sure 
about the removing identifying information part is such a good idea, since 
the best way for people to trust a system is to talk to people that have 
used it before.  Or do we just want the information to filter through the 
asterisk-users list?



--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Getting True ANI not Caller ID


[EMAIL PROTECTED] named]# grep ANI /home/software/asterisk/asterisk-1.2/doc/*
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLERANI} 
* Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLINGANI2} 
   * Caller ANI2 (PRI channels)
/home/software/asterisk/asterisk-1.2/doc/README.variables:${ANI2} 
* The ANI2 Code provided by the network on the incoming 
call.

[EMAIL PROTECTED] named]#


Steve Totaro wrote:
I am trying get true ANI from my provider into asterisk.  I have found a few patches that supposedly accomplish this on Mantis and were committed to CVS sometime mid last year. 
 
My question is, would this have been included in stable 1.2.5 or do have to patch it?  There is very little info on this issue so any help or tips would be appreciated.
 
Thanks,

Steve Totaro




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[Asterisk-Users] Asterisk Failover without SER

2006-03-24 Thread Bryan Mahin









Hello all, I first want to thank everyone for all your
contributions. Ive building an asterisk system for a month or so now and
without everyone in the online asterisk community I wouldnt have made it
this far yet. Thanks! ok, mushiness out of the way.. :)



I am looking for a failover and ultimately a load balancing
asterisk solution. Ive done a good bit of research and I havent
really found any information for implementing an Asterisk only failover or load
balancing solution. Everyone seems to use SER along with asterisk to accomplish
this goal. SER with asterisk may be in my future, but for now I need to get
this system up and running. 



Ive setup heartbeat (ultramonkey), and are able to
take my primary box offline and have the second machine take over, but it isnt
working in regards to asterisk. I cant register phones to the virtual
ip. I can ssh into the virtual ip but my soft phones wont register. I get the
following error. Is this normal?



Does anyone have any experience with this sort of setup
without the use of SER?

Bryan Mahin











Rediscover Personal Servicewith UNETA
Please visit us @ www.uneta.com
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Re: [Asterisk-Users] Maximum Queue Name Length

On 3/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
 Do queue names have a max length? I have a queue named 'oneeighty_techsupp' 
 in queues.conf, and a 'show queues' returns a truncated queue name. Is that 
 just a display bug, or do queues names have a max length of 12?

 demeter*CLI show queues
 oneeighty_te has 0 calls (max unlimited) in 'rrmemory' strategy (0s 
 holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
  Agent/80014255 (Not in use) has taken no calls yet
  Agent/80014257 (Not in use) has taken no calls yet

 I thought the max length of a context name was 16...?


struct ast_call_queue {
ast_mutex_t lock;
char name[80];  /*! Name */
char moh[80];   /*! Music On Hold class to be used */
char announce[80];  /*! Announcement to play when
call is answered */

 If the name is getting trunc'd it's probably from the printf on the
display itself and not that the queue name itself is getting trunc'd.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] Snom 360 problems

2006-03-24 Thread Brian Kennedy


Anyone have a Snom they're happy with?   How did you manage that?   :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841
~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen 
between any two.  Also everything, except the snom, is working sweetly.


The main problem is ECHO.. awful echo and only on the Snom.  When using 
a Zap line or to another sip phone.  I've tweaked the * for echo and 
managed to only create echo for everyone else, pounded the settings in 
the Snom trying to find anything, and updated the firmware to 
Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 v3.36 
after noticing a changelog that sounded like it may have related to 
echo.  Not even a slight reduction in echo so far.


A second serious problem is Call join.   Even with Call join on Xfer (2 
calls) OFF if the user is doing a transfer of one call when a second 
starts ringing the 2 callers get bridged, no transfer.  Really nice, now 
I have two customers talking to each other with no clue what's going on 
and neither gets who they were trying to reach.


Any ideas on what I can try next?

Thanks...
...Brian
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Re: [Asterisk-Users] Changing codec.

I know the answer.  The answer is NO!  Asterisk does not support 
changing the codec during a call.  It also does not support changing the 
codec on an INCOMING call.  Of course, as you know by reading 
README.variables, SIP_CODEC can force a specific codec on an OUTGOING call.


Wai Wu wrote:

Ouch, Come on! One must know.



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Wai Wu

*Sent:* Thursday, March 23, 2006 4:25 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Changing codec.

Hi,

 

Is there a way to tell Asterisk to change the codec being used in the 
middle of an IVR script?





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Re: [Asterisk-Users] getting your own phone number


Matt wrote:

SURE!
800-444-
804-883-2001
800-437-7950
866-692-6447
:)

On 3/24/06, mike webb [EMAIL PROTECTED] wrote:

is there a number is the U.S that you can dial where a computer will
reply with the phone number your calling from.
carrier is sbc if that makes any difference.


Pretty much every central office switch has something implemented to do 
that, but there is no standard number.  I've seen CO's where that 
function is provided via 311 (or something similar), making it 
inaccessible from outside the serving region. In other cases, its a 
random seven digit phone number.


You can try calling your repair number and asking. Some do provide that 
number when asked appropriately.


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Re: [Asterisk-Users] Re: How to nice agi scripts?


setpriority(0, 0, 20);

This is for Perl, of course.

Benny Amorsen wrote:

RS == Roger Schreiter [EMAIL PROTECTED] writes:


RS Is there any mean to let AGI scripts run in a lower priority
RS (except starting a new shell from the a short initial AGI script)?

You can start the script with renice 15 $$, or whichever value you
prefer. If the hickup happens because of the process startup, this
will not help, but at least it is easy to try out.


/Benny


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RE: FW: [Asterisk-Users] Asterisk Users

2006-03-24 Thread Bob McDowell


The only reason I recommended that was to protect the privacy of those
on that list.  I personally do not want a bunch of cold calls from
asterisk 'dealers' just because I chose to implement that product.  Such
a list of users would make a tempting target for marketing uses...

But either way, a list would be a great addition.  It would go a long
way toward debunking the FUD that usually accompanies a product of this
type.  And with Asterisk it's worse because it gets Linux FUD as well as
VoIP FUD.


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, March 24, 2006 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] Asterisk Users

 Perhaps a page on the wiki would work?  We could set the ground rules
 similar to other industries:  no names, nothing more defining than a
 region, the number of units, etc.  Would that be useful?

 For example, I can describe this organization as a security company in

 Southwest Missouri using asterisk with 60 sets and 16 lines.

 When you strip off my name and email, it gets a little less certain
 who I am talking about...


 Bob McDowell

I like the idea of having the information on the wiki, makes it simpler
for everyone to see just how well the project is doing.  I'm not sure
about the removing identifying information part is such a good idea,
since the best way for people to trust a system is to talk to people
that have used it before.  Or do we just want the information to filter
through the asterisk-users list?


--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Call terminated after 60 seconds

For one thing, don't use the r option to dial.  It can hide major 
problems.  If you don't hear ringing without using r then you have 
massive problems.


Asterisk wrote:

Nope,

It's not a firewall problem.
I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.
It replaces the inside IP adresses from the * server in the SIP frames 
by the outside IP adress and creates pinholes for the udp streams.


I have several SIP connections (SIPphone, SIPGate, IPtel, Bugetphone 
...) and only this one is problematic. 


Andre


- Oorspronkelijk Bericht -
*Onderwerp: *Re: [Asterisk-Users] Call terminated after 60 seconds
*Afzender:  *Franc esco Peeters (Asterisk) [EMAIL PROTECTED]
*Aan: *Asterisk [EMAIL PROTECTED],Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
*CC: *[EMAIL PROTECTED]
asterisk-users@lists.digium.com
*Datum: *24-03-2006 12:18


On Fri, March 24, 2006 12:01, Asterisk said:
 
 
  Hello,
 
  I switched from my PSTN provider to a voip provider. (Voicedata in
  the Netherlands)
 From the moment i switched all inbound calls are terminated after
  aproximatly 1 minute.
  The provider tells me it's not their issue since I have no other
  configuration than all their other users.
 
  What can I do.
 
  I removed all asterisk functionality by forwarding the inboud ca ll
  directly to a local phone
  ; Inbound voicedata context
  ;
  [from-voicedata]
  exten = ${VOICEDATACIDNUM},1,NoOp(From Voicedata)
  exten = ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr)
  ; end of context
  Regards,
 
  Andre Vink
 

Check whether your firewall has a fixed UDP timeout set at 60 seconds...
That solved my problem... ;-)
(Together with activating SIP/VoIP support)

-- 
F Peeters

PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
2 Sweex HFC-PCI cards




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[Asterisk-Users] RE: Asterisk Failover without SER

2006-03-24 Thread Bob McDowell

Go through the archives (or your own inbox) for a very, very thorough
set of conversations that just passed this way only a week or two ago.
There are a few key people working on this type of 'HA' solution and
they're pretty close to making it work.  They have already identified
the key issues that need resolving...


Bob McDowell


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan
Mahin
Sent: Friday, March 24, 2006 11:56 AM
To: Asterisk-Users
Subject: [Asterisk-Users] Asterisk Failover without SER



Hello all, I first want to thank everyone for all your contributions.
I've building an asterisk system for a month or so now and without
everyone in the online asterisk community I wouldn't have made it this
far yet. Thanks! ...ok, mushiness out of the way.. :)



I am looking for a failover and ultimately a load balancing asterisk
solution. I've done a good bit of research and I haven't really found
any information for implementing an Asterisk only failover or load
balancing solution. Everyone seems to use SER along with asterisk to
accomplish this goal. SER with asterisk may be in my future, but for now
I need to get this system up and running.



I've setup heartbeat (ultramonkey), and are able to take my primary box
offline and have the second machine take over, but it isn't working in
regards to asterisk. I can't register phones to the virtual ip. I can
ssh into the virtual ip but my soft phones wont register. I get the
following error. Is this normal?



Does anyone have any experience with this sort of setup without the use
of SER?

Bryan Mahin





Rediscover Personal Service with UNETA

Please visit us @ www.uneta.com




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RE: [Asterisk-Users] Asterisk Failover without SER

2006-03-24 Thread Bryan Mahin









Well, I should say Sporadically I
can register to the virtual ip. Other times I cant. 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan Mahin
Sent: Friday, March 24, 2006 12:56
PM
To: Asterisk-Users
Subject: [Asterisk-Users] Asterisk
Failover without SER





Hello all, I first want to thank everyone for all your
contributions. Ive building an asterisk system for a month or so now and
without everyone in the online asterisk community I wouldnt have made it
this far yet. Thanks! ok, mushiness out of the way.. :)



I am looking for a failover and ultimately a load balancing
asterisk solution. Ive done a good bit of research and I havent
really found any information for implementing an Asterisk only failover or load
balancing solution. Everyone seems to use SER along with asterisk to accomplish
this goal. SER with asterisk may be in my future, but for now I need to get
this system up and running. 



Ive setup heartbeat (ultramonkey), and are able to
take my primary box offline and have the second machine take over, but it
isnt working in regards to asterisk. I cant register phones to
the virtual ip. I can ssh into the virtual ip but my soft phones wont register.
I get the following error. Is this normal?



Does anyone have any experience with this sort of setup
without the use of SER?

Bryan Mahin











Rediscover Personal Servicewith UNETA
Please visit us @ www.uneta.com

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RE: [Asterisk-Users] Asterisk Failover without SER

2006-03-24 Thread William Boehlke




You can do a version of failover with phones that support a 
backup registrar. They will repoint themselves to a second server 
then.



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan 
MahinSent: Friday, March 24, 2006 9:56 AMTo: 
Asterisk-UsersSubject: [Asterisk-Users] Asterisk Failover without 
SER


Hello all, I first want to thank 
everyone for all your contributions. Ive building an asterisk system for a 
month or so now and without everyone in the online asterisk community I wouldnt 
have made it this far yet. Thanks! ok, mushiness out of the way.. 
:)

I am looking for a failover and 
ultimately a load balancing asterisk solution. Ive done a good bit of research 
and I havent really found any information for implementing an Asterisk only 
failover or load balancing solution. Everyone seems to use SER along with 
asterisk to accomplish this goal. SER with asterisk may be in my future, but for 
now I need to get this system up and running. 

Ive setup heartbeat (ultramonkey), 
and are able to take my primary box offline and have the second machine take 
over, but it isnt working in regards to asterisk. I cant register phones to 
the virtual ip. I can ssh into the virtual ip but my soft phones wont register. 
I get the following error. Is this normal?

Does anyone have any experience with 
this sort of setup without the use of SER?
Bryan 
Mahin






Rediscover Personal Servicewith 
UNETA
Please visit us @ 
www.uneta.com
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[Asterisk-Users] RE: Snom 360 problems

2006-03-24 Thread Usman Tahir


Hi Brian,

For the conf on Xfer issue, use the latest beta
http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin

Regards,

-
Usman Tahir
snom technology AG 
Gradestraße 46 
D-12347 Berlin. 
Tel: +49 30 398330 
Fax: +49 30 39833111 
[EMAIL PROTECTED]
www.snom.com  

This e-mail may contain confidential and/or privileged information. If you
are not the intended recipient (or have received this e-mail in error)
please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen
enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail
irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und
vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte
Weitergabe dieser Mail sind nicht gestattet.

-

Date: Fri, 24 Mar 2006 12:41:26 -0500
From: Brian Kennedy [EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom 360 problems
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Anyone have a Snom they're happy with?   How did you manage that?  :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar 


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[Asterisk-Users] VoIP QoS monitoring and failover re-routing

2006-03-24 Thread Tristram Graham

Hi,

I am looking at a project which requires VoIP QoS monitoring and failover
re-routing to PSTN, without dropping the call ideally. While I have hardware
solutions available such as the Quintum Tenor series, I see no reason why
Asterisk can't have this feature with some effort obviously. Plus I note it
is on the wish list at http://www.voip-info.org/wiki/view/Asterisk+Wishlist
(last item under General).

As part of my investigation, I was wondering if anyone has any information
on projects or tools for Asterisk that move towards this feature? My initial
searches has only revealed commercial tools, academic studies and manual
tools for QoS. But nothing directly related to this feature in Asterisk.
Either way, I'd certainly be happy to help with any existing efforts or
seeing if time can be found to implement into Asterisk from scratch.

Thanks, Tris

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Re: FW: [Asterisk-Users] Asterisk Users

2006-03-24 Thread Gary Richardson

You wouldn't have to post anonymously -- only if it makes you feel better.

I could have really used such a resource in January -- Digium's list
of success stories is a little thin.

On 3/24/06, Aaron Daniel [EMAIL PROTECTED] wrote:
  Perhaps a page on the wiki would work?  We could set the ground rules
  similar to other industries:  no names, nothing more defining than a
  region, the number of units, etc.  Would that be useful?
 
  For example, I can describe this organization as a security company in
  Southwest Missouri using asterisk with 60 sets and 16 lines.
 
  When you strip off my name and email, it gets a little less certain who
  I am talking about...
 
 
  Bob McDowell

 I like the idea of having the information on the wiki, makes it simpler
 for everyone to see just how well the project is doing.  I'm not sure
 about the removing identifying information part is such a good idea, since
 the best way for people to trust a system is to talk to people that have
 used it before.  Or do we just want the information to filter through the
 asterisk-users list?


 --
 Aaron Daniel
 Computer Systems Technician
 Sam Houston State University
 [EMAIL PROTECTED]
 (936) 294-4198
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[Asterisk-Users] Realtime Agents

2006-03-24 Thread Douglas Garstang

In short, does this work yet?

ie putting agents into Realtime. Can't find any info on it...

Thanks,
Doug.
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RE: FW: [Asterisk-Users] Asterisk Users

I see your reasoning on that one.  Perhaps it is best to leave the 
information about the asterisk-users list so they can contact us.  I'm 
just worried about the idea that people aren't going to want to subscribe 
to a 4000 message a month list just to find out about a system. 
Especially with the emails that come through here, setting up the system 
will seem pretty daunting, probably even discouraging.  Any ideas on that?


Aaron

On Fri, 24 Mar 2006, Bob McDowell wrote:



The only reason I recommended that was to protect the privacy of those
on that list.  I personally do not want a bunch of cold calls from
asterisk 'dealers' just because I chose to implement that product.  Such
a list of users would make a tempting target for marketing uses...

But either way, a list would be a great addition.  It would go a long
way toward debunking the FUD that usually accompanies a product of this
type.  And with Asterisk it's worse because it gets Linux FUD as well as
VoIP FUD.


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Friday, March 24, 2006 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] Asterisk Users


Perhaps a page on the wiki would work?  We could set the ground rules
similar to other industries:  no names, nothing more defining than a
region, the number of units, etc.  Would that be useful?

For example, I can describe this organization as a security company in



Southwest Missouri using asterisk with 60 sets and 16 lines.

When you strip off my name and email, it gets a little less certain
who I am talking about...


Bob McDowell


I like the idea of having the information on the wiki, makes it simpler
for everyone to see just how well the project is doing.  I'm not sure
about the removing identifying information part is such a good idea,
since the best way for people to trust a system is to talk to people
that have used it before.  Or do we just want the information to filter
through the asterisk-users list?


--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Queue Period-Announce

2006-03-24 Thread Wes Baehr

In my queue, I have defined a periodic announcement with a message that goes 
something like if you would like to leave a voice message now, please press 
1 However, when a user presses 1 *during* the message, the playback stops 
and the user still remains in the queue (listening to music on hold). The same 
also applies during other periodic announcements (such as there are x calls 
waiting etc).

The caller will be transferred to the queue context if they press 1 while 
listening to MOH.

Is this by design or is it a bug, and is there a workaround? (besides changing 
my prompt)

Thanks! :)


Wes Baehr
Ability Business Computing, Ltd.
Office: 330.882.0455 x25     Cell: 330.882.0455 x35
[EMAIL PROTECTED]
 

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RE: [Asterisk-Users] Snom 360 problems

2006-03-24 Thread Guido Hecken

 Anyone have a Snom they're happy with?   How did you manage that?  :)
 
 I have a system of:
 
 Asterisk 1.2.3
 2 Wildcard TDM400P  Rev I and E/F
 1 Snom 360 + sidecar
 ~15 Sipura/Linsys SPA-841
 ~15 Grandstream 101
 
 Everything (currently) is on the same network, not a router to be seen
 between any two.  Also everything, except the snom, is working sweetly.
 
 The main problem is ECHO.. awful echo and only on the Snom.  When using
 a Zap line or to another sip phone.  I've tweaked the * for echo and
 managed to only create echo and piss everyone else off, pounded the
 settings in the Snom trying to find anything, and updated the firmware
 to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2
 v3.36 after noticing a changelog that sounded like it may have related
 to echo.  Not even a slight reduction in echo so far.
 
 A second serious problem is Call join.   Even with Call join on Xfer (2
 calls) OFF if the user is doing a transfer of one call when a second
 starts ringing the 2 callers get bridged, no transfer.  Really nice, now
 I have two customers talking to each other with no clue what's going on
 and neither gets who they were trying to reach.
 
 Any ideas on what I can try next?

This firmware works well for us: snom360-SIP 4.1 available here:
http://snom.com/download/share/snom360-4.1-SIP-j.bin
No echo and overall voice quality is excellent.

Did you check the codecs on the snom and on asterisk (sip.conf)?
Is Silence Suppression off on the snom?
If you would post your config (under settings on the snom) we could have a
closer look in the problem.

Regards, 

Guido
 
gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany

fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]


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RE: [Asterisk-Users] Asterisk Failover without SER