Re: [asterisk-users] BLF trouble

[Lars Bensmann]

Thanks a lot. Works like a charm.

-- 
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  -- Katrin Rahms

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Re: [asterisk-users] BLF trouble

[Lars Bensmann]

On Tue, Dec 18, 2007 at 02:10:25PM +0100, Lars Bensmann wrote:
 I will make some more tests and gather some CLI output.

han*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
 [EMAIL PROTECTED] : SIP/pioSIP/pio-mobi  
State:IdleWatchers  2
 [EMAIL PROTECTED] : SIP/sekretariat   
State:IdleWatchers  1

-- Executing [EMAIL PROTECTED]:43] Dial(SIP/sekretariat-08225e70, 
SIP/pioSIP/pio-mobil|15|tT) in new stack
-- Called pio
 Extension Changed 13 new state Ringing for Notify User pio
 Extension Changed 13 new state Ringing for Notify User sekretariat
-- Called pio-mobil
-- SIP/pio-0820eff8 is ringing
-- SIP/pio-mobil-08235218 is ringing

han*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last 
Message
x.y.z.240pio-mobil   240bac96767  00102/0  alaw  No   Init: 
INVITE
x.y.z.235pio 052c87ac0dd  00102/0  alaw  No   Init: 
INVITE
x.y.z.239sekretaria  d01f84e350f  00101/30922  alaw  No   Rx: INVITE

han*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
 [EMAIL PROTECTED] : SIP/pioSIP/pio-mobi  
State:Ringing Watchers  2
 [EMAIL PROTECTED] : SIP/sekretariat   
State:IdleWatchers  1

  == Spawn extension (macro-gbit-std-exten, s, 43) exited non-zero on 
'SIP/sekretariat-08225e70'
 Extension Changed 13 new state Idle for Notify User pio
 Extension Changed 13 new state Idle for Notify User sekretariat


Why is extension 12 not updated as being in use? When extension 13 calls 12 the
situation is exactly the same, but this time '[EMAIL PROTECTED]' is Ringing 
and '[EMAIL PROTECTED]'
is Idle.

Is this some configuration error on my part? But I really have no idea where to
look for this.

Any help is appreciated,
Lars

-- 
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Re: [asterisk-users] Asterisk.NET API --help required

[Lee Jenkins]

srinivas Antarvedi wrote:
 Hello all,
 
 Here is the requirement from my side
 to  use Asterisk.NET API  to  generate
 an automated call  (outgoing)  from asterisk
 and then link to one of the extensions which
 plays a sound file for the callee.
 
 For this i have worked out in the follwing way
 
 1)modified manager.conf to facilitate this API to talk to asterisk
 2)used the command Originate to call a Registered user under
asterisk and when the user answers the phone it plays whatever
i put against the extension..
 
 But my exact requirement is like this
 
 1)Call to the user
 2)if answers connect him to the extension provided in the extensions.conf
 3)if the user didnt lift the phone within the deault timeout period(30 sec)
 4)if the user cancels the phone (Congestion case)
 5)if the user not registerd to the(unreachable case)
 
 to trace the cases of 3, 4, 5 how should i follow the API
 I got confused with originate action,orginate sucess event , originate 
 failure event
 

http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx

Download the Manager API Testing Utility.  I wrote to help me with a software 
program that I was writing that used the Manager API heavily.  Allows you to 
view the AMI activity, send commands, etc outside of your dev environment. 
Helped me a lot and its fee to use.

You can also get it on:
http://www.voip-info.org/wiki/view/Asterisk+GUI

Wish I had more time to do Asterisk related development, its a lot of fun...

-- 
Warm Regards,

Lee

If I don't see you around here, I'll see you around, hear?

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Re: [asterisk-users] All trunk are busy please try your call again later

[Lolu Gbenga]

Good Day

Find attached the relevant portions of the asterisk CLI.

Please,which portion of the extension .conf should i send ?

It is connected via RJ 45 connector to an E1 modem to the telco company.

I use E1 link.

I will appreciate your reply.

Best Regards


On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote:

 lolu,
 sounds more like a telco/itsp problem then *.
 I would
tcpdump -i eth0 port 5060
 to make sure it is actually going out... change 5060 if you have changed
 your port to your itsp, of course.
 to see what is going on as well as the other debugging notes mentioned
 in this thread.
 daveC

 Lolu Gbenga wrote:
  Good Day all
 
  Please I am having some issues on my voip asterisk server
 
  I make internal calls on extensions configured ie extension 192 can
  call extension 195 etc
 
  But each time i try to make calls outside the extension ie calling a
  GSM or an external line ,i always hear this response all trunk calls
  are busy please try your call again later
 
  Please how can i resolve this problem .
 
  I will appreciate your response.
 
  Best Regards
 
  Success
 
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 --
 My wife's sister is in California.
 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

 WorldWideVideoPhones.com
 856.380.0894




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SIP SHOW PEERS 

Name/username  HostDyn Nat ACL Port Status
7871/7871 (Unspecified)D  0Unmonitored

...
...

7874/7874 (Unspecified)D  0Unmonitored
108 sip peers [108 online , 0 offline]
Verbosity is at least 3


ZAP SHOW CHANNELS

 Chan Extension  Context Language   MusicOnHold 
 pseudodefault en 
  1default en 
  2default en 
  

ZAP SHOW CHANNELS
Description  Alarms IRQbpviol CRC4  

T4XXP (PCI) Card 0 Span 1OK 0  0  0 

T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0  0  0 

T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0  0  0 

T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0  0  0 

ZTDUMMY/1 1  UNCONFIGUR 0  0  0 

Verbosity is at least 3
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Re: [asterisk-users] All trunk are busy please try your call again later

[Lolu Gbenga]

Thanks
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port and
port number.

I will appreciate your  reply.

Best Regards


On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote:

 lolu,
 sounds more like a telco/itsp problem then *.
 I would
tcpdump -i eth0 port 5060
 to make sure it is actually going out... change 5060 if you have changed
 your port to your itsp, of course.
 to see what is going on as well as the other debugging notes mentioned
 in this thread.
 daveC

 Lolu Gbenga wrote:
  Good Day all
 
  Please I am having some issues on my voip asterisk server
 
  I make internal calls on extensions configured ie extension 192 can
  call extension 195 etc
 
  But each time i try to make calls outside the extension ie calling a
  GSM or an external line ,i always hear this response all trunk calls
  are busy please try your call again later
 
  Please how can i resolve this problem .
 
  I will appreciate your response.
 
  Best Regards
 
  Success
 
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 --
 My wife's sister is in California.
 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

 WorldWideVideoPhones.com
 856.380.0894




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Re: [asterisk-users] ip phone suggestion for Asia?

[GNUbie]

On Dec 20, 2007 12:33 PM, d tbsky [EMAIL PROTECTED] wrote:

 Hi:
   i am surveying ip phones for our company. we will use them with
 asterisk.
   we have office in taiwan, hong kong,singapore and china.
   cisco and polycom are too expensive for us.
   we try several china brand ip phones. they are all cheap and
 some of them have good quality. but most of them won't offer future
 firmware
 support, which we think it's important for ip phones.
   searching in the mail list, we found aastra is good, but they don't sale
 to
 asia. grandstream looks good also.there are many grandstream users in the
 list,
 can someone share any good or bad experience about grandstream today?
   if there are other good choice, please tell us!!
   thanks a lot for your help!!


Try getting the Aztech IP150 
http://www.aztech.com.sg/ip_telephony/ip150.html which is based on the SNOM
300.

Regards,

GNUbie
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Re: [asterisk-users] All trunk are busy please try your call again later

[Lolu Gbenga]

Hi Steve
Am connected  to the telco  through an E1 link using modem(Watson 5  modem
SDHSL 1 PAIR schmid telecommunications).The MODEM is connected to the
asterisk box through RJ 45 to the asterisk box end  and serial connector to
the modem end .
Which portion of the extension conf should i post ?
Thanks

On Dec 18, 2007 12:03 PM, Steve Totaro [EMAIL PROTECTED] wrote:

 Lolu Gbenga wrote:
  Good Day all
 
  Please I am having some issues on my voip asterisk server
 
  I make internal calls on extensions configured ie extension 192 can
  call extension 195 etc
 
  But each time i try to make calls outside the extension ie calling a
  GSM or an external line ,i always hear this response all trunk calls
  are busy please try your call again later
 
  Please how can i resolve this problem .
 
  I will appreciate your response.
 
  Best Regards
 
  Success
 

 You need to at least post some verbose from the console and explain how
 you are connecting to the PSTN.  It would greatly help if you included
 the relevant portions of your extensions.conf.

 Thanks,
 Steve Totaro

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Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

lolu,
while you are making the call., capture and post your CLI output
... this is easy to do since you are using putty.

login to your pbx and start asterisk, use the below command:

# asterisk -vvvr

then make the call. hilite the text on the putty terminal and paste it
into the body of the email to the list... 
sorry if I'm making these instruction too basic...

pbv01*CLI
 -- Executing [EMAIL PROTECTED]:1] Wait("SIP/202-b753da18", "1") in new
stack
 -- Executing [EMAIL PROTECTED]:2] Answer("SIP/202-b753da18", "") in
new stack
 -- Executing [EMAIL PROTECTED]:3] NoOp("SIP/202-b753da18", "DEBUG:
CALLERID=") in new stack
 -- Executing [EMAIL PROTECTED]:4] Notify("SIP/202-b753da18",
"800202|x202|300/192.168.15.100") in new stack
 -- Notify: sending '800202|x202|300' to 192.168.15.100:4
 -- Executing [EMAIL PROTECTED]:5] AGI("SIP/202-b753da18",
"agi-callpop4.sh||red") in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-callpop4.sh
 -- AGI Script agi-callpop4.sh completed, returning 0
 -- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b753da18", "AGISTATUS
is FAILURE") in new stack
 -- Executing [EMAIL PROTECTED]:7] NoOp("SIP/202-b753da18", "DEBUG:
EXTEN=300") in new stack
 -- Executing [EMAIL PROTECTED]:8] Dial("SIP/202-b753da18",
"SIP/300|15|rt") in new stack
 -- Called 300
 -- SIP/300-09e062e8 is ringing
 == Spawn extension (local-sip, 300, 8) exited non-zero on
'SIP/202-b753da18'

daveC





Lolu Gbenga wrote:
Thanks 
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port
and port number.
  
I will appreciate your reply.
  
Best Regards 
  
  
  On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED]
wrote:
  lolu,
sounds more like a telco/itsp problem then *.
I would
 tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned

in this thread.
daveC


Lolu Gbenga wrote:
 Good Day all

 Please I am having some issues on my voip asterisk server

 I make internal calls on extensions configured ie extension 192
can

 call extension 195 etc

 But each time i try to make calls outside the extension ie calling
a
 GSM or an external line ,i always hear this response "all trunk
calls
 are busy please try your call again later"


 Please how can i resolve this problem .

 I will appreciate your response.

 Best Regards

 Success




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--
My wife's sister is in California.

I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.503 / Virus Database: 269.17.2 - Release Date: 12/14/2007 12:00 AM
  


-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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Re: [asterisk-users] All trunk are busy please try your call again later

[Steve Totaro]

What is the output of ztconfig from the Linux command line?  What does 
your zaptel.conf and zapata.conf look like?  What is the relevant part 
of extensions.conf (the dialout section that fails).  Also from the CLI, 
it would be most helpful to post the output you get when dialing out 
fails.  I don't think it is a network issue at all, I think your configs 
need some work.

Thanks,
Steve Totaro

Lolu Gbenga wrote:
 Good Day

 Find attached the relevant portions of the asterisk CLI.

 Please,which portion of the extension .conf should i send ?

 It is connected via RJ 45 connector to an E1 modem to the telco company.

 I use E1 link.

 I will appreciate your reply.

 Best Regards


 On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] 
 mailto:[EMAIL PROTECTED]  wrote:

 lolu,
 sounds more like a telco/itsp problem then *.
 I would
tcpdump -i eth0 port 5060
 to make sure it is actually going out... change 5060 if you have
 changed
 your port to your itsp, of course.
 to see what is going on as well as the other debugging notes mentioned
 in this thread.
 daveC

 Lolu Gbenga wrote:
  Good Day all
 
  Please I am having some issues on my voip asterisk server
 
  I make internal calls on extensions configured ie extension 192 can
  call extension 195 etc
 
  But each time i try to make calls outside the extension ie calling a
  GSM or an external line ,i always hear this response all trunk
 calls
  are busy please try your call again later
 
  Please how can i resolve this problem .
 
  I will appreciate your response.
 
  Best Regards
 
  Success
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 

 --
 My wife's sister is in California.
 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

 WorldWideVideoPhones.com
 856.380.0894







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Re: [asterisk-users] [VOIP-Users-Conference] Re: Digium: as of this a.m., one million Asterisk downloads this year

[Dean Collins]

lol - yep when news of this first broke I thought thats actually a very
good idea to have implemented, though it sounds the way Trixbox
implemented it may have been unsecure.

Maybe someone else can come up with a better way of implementing this.

If the data was all randomised there's no harm in doing this;


some basic infomration like;
Hours of uptime
Reboots
Number of extensions
Number of calls
Number of minutes

Make it totally voluntary and this would be a good thing to Digium for
them to be able to be more aware of whats happening out there...apart
from users downloaded the application 20 kazillion times this year.

Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeek
(randulo)
Sent: Thursday, 20 December 2007 8:49 AM
To: VOIP Users Conference
Subject: [VOIP-Users-Conference] Re: Digium: as of this a.m., one
million Asterisk downloads this year


Now if Digium had a voluntary call home script we'd know how many
people actually are using it! The way traditional software companies do
this would be a once a month login that did a GET with a current
version. I actually think it'd be a good idea (assuming it's
voluntary) to include such a script in asterisk.

On Dec 20, 2:51 am, Michael Graves [EMAIL PROTECTED] wrote:
 http://blogs.zdnet.com/ip-telephony/?p=2903

--~--~-~--~~~---~--~~
Your participation in the conference is always appreciated! Please try
to be there live when it happens.

You received this message because you are subscribed to the Google
Groups Asterisk Users Conference group.
To post to this group, send email to
[EMAIL PROTECTED]
To unsubscribe from this group, send email to
[EMAIL PROTECTED]
For more options, visit this group at
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[asterisk-users] Cisco 7961 new firmware stops reading configuration files

[Chad Osmond]

Hello,
 
I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have
recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front
of the phone and also to hopefully resolve some issues with the phones
not registering after a long period.

Once we upgraded the phones now display Error Verifying Config Info in
the Status messages and will not process the configuration file.

To make a change on the phone I have to downgrade to 8.2.2R4 and change
the configuration, and then upgrade to 8.3.2R1, which is a bit of a
pain.

The tftp logs indicate that the phones is getting the correct
SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be
the issue.

The Wiki pages for 79x1 indicate that it's a known issue, has anyone
managed to get past the issue?
I tried logging a call with Cisco TAC, but they're giving the We don't
support SIP on anything other then CME...

Thanks,

Chad


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[asterisk-users] put fxo channel before E1 channel?

[d tbsky]

hi:
 my system has one 4-port fxo card and one 2-port E1 card.
 for some reason, i like to place fxo as channel 1-4, and E1 use the
rest channels (5-66).
 i modify zaptel.conf, and ztcfg -vv is happy. but asterisk seems
not happy with
this configuration. it still want channel 16 as D-channel, in my case
the D-channel should
be 20. i don't know if this is a limit of asterisk. i play some
parameters in zapata.conf
like trunkgroup. but i still can not get it work.
 any suggestion? or it is not allowed?
 thanks for your help!!

Regards,
tbskyd

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Re: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files

[Anciso, Roy]

Chad,
You might want to upgrade to the latest firmware. I have 7961g on
8-3-3SR2S and works very well.  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad
Osmond
Sent: Thursday, December 20, 2007 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cisco 7961 new firmware stops
readingconfiguration files

Hello,
 
I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have
recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front
of the phone and also to hopefully resolve some issues with the phones
not registering after a long period.

Once we upgraded the phones now display Error Verifying Config Info in
the Status messages and will not process the configuration file.

To make a change on the phone I have to downgrade to 8.2.2R4 and change
the configuration, and then upgrade to 8.3.2R1, which is a bit of a
pain.

The tftp logs indicate that the phones is getting the correct
SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be
the issue.

The Wiki pages for 79x1 indicate that it's a known issue, has anyone
managed to get past the issue?
I tried logging a call with Cisco TAC, but they're giving the We don't
support SIP on anything other then CME...

Thanks,

Chad


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Re: [asterisk-users] All trunk are busy please try your call again later

[Lolu Gbenga]

Hi all,
I am grateful for our contribution so far .

I followed dave advise and i have the attached file using the aterisk -r
when a call is made.

I attached two files.

One of the attached file is for the external call,which replied with the
PROBLEM all trunks are busy now,please try your call again later.

The second attachment is when i made internal calls and the phone rang.

Please,i will be expecting your replies for further directions.

Best Regards


On Dec 20, 2007 2:58 PM, Steve Totaro  [EMAIL PROTECTED]
wrote:

 What is the output of ztconfig from the Linux command line?  What does
 your zaptel.conf and zapata.conf look like?  What is the relevant part
 of extensions.conf (the dialout section that fails).  Also from the CLI,
 it would be most helpful to post the output you get when dialing out
 fails.  I don't think it is a network issue at all, I think your configs
 need some work.

 Thanks,
 Steve Totaro

 Lolu Gbenga wrote:
  Good Day
 
  Find attached the relevant portions of the asterisk CLI.
 
  Please,which portion of the extension .conf should i send ?
 
  It is connected via RJ 45 connector to an E1 modem to the telco company.
 
  I use E1 link.
 
  I will appreciate your reply.
 
  Best Regards
 
 
  On Dec 18, 2007 4:02 PM, dave cantera  [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED]  wrote:
 
  lolu,
  sounds more like a telco/itsp problem then *.
  I would
 tcpdump -i eth0 port 5060
  to make sure it is actually going out... change 5060 if you have
  changed
  your port to your itsp, of course.
  to see what is going on as well as the other debugging notes
 mentioned
  in this thread.
  daveC
 
  Lolu Gbenga wrote:
   Good Day all
  
   Please I am having some issues on my voip asterisk server
  
   I make internal calls on extensions configured ie extension 192
 can
   call extension 195 etc
  
   But each time i try to make calls outside the extension ie calling
 a
   GSM or an external line ,i always hear this response all trunk
  calls
   are busy please try your call again later
  
   Please how can i resolve this problem .
  
   I will appreciate your response.
  
   Best Regards
  
   Success
  
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Re: [asterisk-users] All trunk are busy please try your call again later

[Lolu Gbenga]

Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT  USING asterisk -vvvr command for EXTERNAL calls that
gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.

Verbosity is at least 3
-- Executing Macro(SIP/7871-f813, dialout-trunk|1|018774957||)
 in new sta ck
-- Executing GotoIf(SIP/7871-f813, 1?3:2) in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro(SIP/7871-f813, user-callerid) in new stack
-- Executing Set(SIP/7871-f813, AMPUSER=7871) in new stack
-- Executing Set(SIP/7871-f813, EMERGENCYCID=7871) in new stack
-- Executing Set(SIP/7871-f813, AMPUSERCIDNAME=7871) in new
 stack
-- Executing GotoIf(SIP/7871-f813, 0?6) in new stack
-- Executing Set(SIP/7871-f813, CALLERID(all)=7871 7871) in
 new stack
-- Executing NoOp(SIP/7871-f813, Using CallerID 7871 7871)
 in new stack
-- Executing Macro(SIP/7871-f813, record-enable|7871|OUT) in
 new stack
-- Executing GotoIf(SIP/7871-f813, 0  0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/7871-f813,
 recordingcheck|20051006-001624|1128554184.
  8) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20051006-001624|1128554184.8: Outbound recording not
 enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(SIP/7871-f813, No recording needed) in new
 stack
-- Executing Macro(SIP/7871-f813, outbound-callerid|1) in new
 stack
-- Executing Set(SIP/7871-f813, USEROUTCID=7871) in new stack
-- Executing GotoIf(SIP/7871-f813, 1?4) in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf(SIP/7871-f813, 0?6) in new stack
-- Executing Set(SIP/7871-f813, CALLERID(all)=7871) in new
 stack
-- Executing GotoIf(SIP/7871-f813, 1?8) in new stack
-- Goto (macro-outbound-callerid,s,8)
-- Executing NoOp(SIP/7871-f813, CallerID set to  7871) in
 new stack
-- Executing Set(SIP/7871-f813, GROUP()=OUT_1) in new stack
-- Executing GotoIf(SIP/7871-f813, 0?108) in new stack
-- Executing Set(SIP/7871-f813, DIAL_NUMBER=018774957) in new
 stack
-- Executing Set(SIP/7871-f813, DIAL_TRUNK=1) in new stack
-- Executing AGI(SIP/7871-f813, fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Removed prefix. New number: 8774957
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set(SIP/7871-f813, OUTNUM=8774957) in new stack
-- Executing Set(SIP/7871-f813, custom=ZAP/1) in new stack
-- Executing GotoIf(SIP/7871-f813, 0?16) in new stack
-- Executing Dial(SIP/7871-f813, ZAP/1/8774957|120|W) in new
 stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing it to SIP/7871-f813
Don't know what to do if second ROSE component is of type 0x6
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto(SIP/7871-f813, s-CHANUNAVAIL|1) in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp(SIP/7871-f813, Dial failed due to
 CHANUNAVAIL) in new s tack
-- Executing Macro(SIP/7871-f813, outisbusy|) in new stack
-- Executing Playback(SIP/7871-f813, all-circuits-busy-now) in
 new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback(SIP/7871-f813, pls-try-call-later) in new
 stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro(SIP/7871-f813, hangupcall) in new stack
-- Executing ResetCDR(SIP/7871-f813, w) in new stack
-- Executing NoCDR(SIP/7871-f813, ) in new stack
-- Executing Wait(SIP/7871-f813, 5) in new stack
-- Executing Hangup(SIP/7871-f813, ) in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
 'SIP/7871-f813'  in macro
 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
 'SIP/7871-f813'  in macro
 'outisbusy'
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
 'SIP/7871-f813'
asterisk1*CLI


 ALSO FIND BELOW THE OUTPUT using asterisk -vvvr command FOR INTERNAL calls
that rang.

Verbosity is at least 3
-- Executing Macro(SIP/7871-bb64, exten-vm|novm|7874) in new
 stack
-- Executing Macro(SIP/7871-bb64, user-callerid) in new stack
-- Executing Set(SIP/7871-bb64, AMPUSER=7871) in new stack
-- Executing Set(SIP/7871-bb64, EMERGENCYCID=7871) in new stack
-- Executing Set(SIP/7871-bb64, AMPUSERCIDNAME=7871) in new
 stack
-- Executing GotoIf(SIP/7871-bb64, 0?6) in new stack
-- Executing Set(SIP/7871-bb64, CALLERID(all)=7871 7871) in
 new stack
-- Executing NoOp(SIP/7871-bb64, Using CallerID 

Re: [asterisk-users] ip phone suggestion for Asia?

[d tbsky]

hi gnubie:
   snom seems has some re-brand ip phones. do they use the same firmware?
if they are the same, i don't understand why snom do this..

Regards,
tbskyd

2007/12/20, GNUbie [EMAIL PROTECTED]:
 On Dec 20, 2007 12:33 PM, d tbsky [EMAIL PROTECTED] wrote:

  Hi:
i am surveying ip phones for our company. we will use them with
 asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco and polycom are too expensive for us.
we try several china brand ip phones. they are all cheap and
  some of them have good quality. but most of them won't offer future
 firmware
  support, which we think it's important for ip phones.
searching in the mail list, we found aastra is good, but they don't sale
 to
  asia. grandstream looks good also.there are many grandstream users in the
 list,
  can someone share any good or bad experience about grandstream today?
if there are other good choice, please tell us!!
thanks a lot for your help!!
 

 Try getting the Aztech IP150
 http://www.aztech.com.sg/ip_telephony/ip150.html which is
 based on the SNOM 300.

 Regards,

 GNUbie

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[asterisk-users] OT: VoIP SLA for SIP trunking - SMEs

[Marco Mouta]

Hi guys,

I know that this could be considered a bit off the topic, I've just posted
this topic at VoIPSEC mailing list but I just thought this could be very
interesting for Asterisk community members so I'm posting it here too.

So the point is for traditional telephony we expect service availability of
99,999% and what about VoIP providers around the world what is currently
available?

I know that some operators are not owners of the infrastructure so this
becomes even harder to warranty the high level requirements that telephony
demands.

The best info and explanation I've found until now is: from bandwidth
at http://www.bandwidth.com/pdf/voip/bandwidth_voip_sla_062105.pdf


It would be interesting to discuss the recommend values needed and available
at SLA agreements:

Max Latency
Max Jitter
Max Packet Lost

Mean Time Between Failures
Mean Time To Repair

Mean Opinion Score at least 4 ?

Service Availability, this point as you probably are aware is very
important, what I just notice a few time a go is that the difference between
99,9% and 99,999% is big!

99,9%-  Max time for Outage during one month is 43,2 minutes ,
considering 30 days per month
99,999% -  Max time for Outage during one month is 0,432 minutes

If any of you around the world is aware of  this values for VoIP SLAs I
would be thankful to exchange and discuss this info.

Thanks in advance.

Best regards,
Marco Mouta

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Re: [asterisk-users] All trunk are busy please try your call again later

[dave cantera]

lolu
I reformated the output so it was easier to understand. I attached the
word document for you.
on the below line:

 --
Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in
new stack
 -- Requested transfer capability:
0x00 - SPEECH
 -- Called 1/8774957
 -- Zap/1-1 is proceeding passing it
to SIP/7871-f813 Don't know what to do if second ROSE component is
of type 0x6


it looks
like this is where it determines it can't proceed... also, there are
many tests along the way... we don't know about the
questions/conditions and if that effects it or not... probably not..

in any case, the question you must answer is 'what is the second
ROSE component'??? and why is of type 0x6???
how is it set and by what component?
hope that moves you closer to the ultimate resolution...
daveC


Lolu Gbenga wrote:
Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls
that gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.
  
Verbosity is at least 3
-- Executing Macro("SIP/7871-f813", "dialout-trunk|1|018774957||")
 in new sta ck
-- Executing GotoIf("SIP/7871-f813", "1?3:2") in new stack

-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/7871-f813", "user-callerid") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSER=7871") in new stack

-- Executing Set("SIP/7871-f813", "EMERGENCYCID=7871") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSERCIDNAME=7871") in new
 stack
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack

-- Executing Set("SIP/7871-f813", "CALLERID(all)="7871" 7871") in
 new stack
-- Executing NoOp("SIP/7871-f813", "Using CallerID "7871" 7871")

 in new stack
-- Executing Macro("SIP/7871-f813", "record-enable|7871|OUT") in
 new stack
-- Executing GotoIf("SIP/7871-f813", "0  0?2:4") in new stack
-- Goto (macro-record-enable,s,4)

-- Executing AGI("SIP/7871-f813",
 "recordingcheck|20051006-001624|1128554184. 8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

  recordingcheck|20051006-001624|1128554184.8: Outbound recording not
 enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/7871-f813", "No recording needed") in new

 stack
-- Executing Macro("SIP/7871-f813", "outbound-callerid|1") in new
 stack
-- Executing Set("SIP/7871-f813", "USEROUTCID=7871") in new stack
-- Executing GotoIf("SIP/7871-f813", "1?4") in new stack

-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
-- Executing Set("SIP/7871-f813", "CALLERID(all)=7871") in new

 stack
-- Executing GotoIf("SIP/7871-f813", "1?8") in new stack
-- Goto (macro-outbound-callerid,s,8)
-- Executing NoOp("SIP/7871-f813", "CallerID set to "" 7871") in

 new stack
-- Executing Set("SIP/7871-f813", "GROUP()=OUT_1") in new stack
-- Executing GotoIf("SIP/7871-f813", "0?108") in new stack
-- Executing Set("SIP/7871-f813", "DIAL_NUMBER=018774957") in new

 stack
-- Executing Set("SIP/7871-f813", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/7871-f813", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

  fixlocalprefix: Removed prefix. New number: 8774957
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/7871-f813", "OUTNUM=8774957") in new stack
-- Executing Set("SIP/7871-f813", "custom=ZAP/1") in new stack

-- Executing GotoIf("SIP/7871-f813", "0?16") in new stack
-- Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in new
 stack
-- Requested transfer capability: 0x00 - SPEECH

-- Called 1/8774957
-- Zap/1-1 is proceeding passing it to SIP/7871-f813
Don't know what to do if second ROSE component is of type 0x6
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'

  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/7871-f813", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/7871-f813", "Dial failed due to

 CHANUNAVAIL") in new s tack
-- Executing Macro("SIP/7871-f813", "outisbusy|") in new stack
-- Executing Playback("SIP/7871-f813", "all-circuits-busy-now") in

 new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/7871-f813", "pls-try-call-later") in new
 stack
-- Playing 'pls-try-call-later' (language 'en')

-- Executing Macro("SIP/7871-f813", "hangupcall") in new stack
-- Executing ResetCDR("SIP/7871-f813", "w") in new stack
-- Executing NoCDR("SIP/7871-f813", "") in new stack

-- Executing Wait("SIP/7871-f813", "5") in new stack
-- Executing Hangup("SIP/7871-f813", "") in new stack
  == Spawn 

Re: [asterisk-users] hi

[Mojo with Horan Company, LLC]

Is it behind a router? either forward the necessary ports to the sip 
phone's internal network ip address using the router, or move the phone 
outside the router to get it an external network (global ip) ;)

Mojo

sandeep.s wrote:
 Hi,
 my sip phone is unreachable for external network(global ip)


 Thanks,
 sandeep.s

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Re: [asterisk-users] How to change sendmail return path

[Mojo with Horan Company, LLC]

Tilghman Lesher wrote:
 On Wednesday 19 December 2007 17:44:15 shadowym wrote:
   
 I had high hopes for this solution for unfortunately it's not working.  Did
 exactly as you specified but return path is still [EMAIL PROTECTED]
 even though [EMAIL PROTECTED] in voicemail.conf :(
 

 Did you restart Sendmail?  It doesn't pick up changes to its config file
 otherwise.

   
And if you modified sendmail.mc instead of sendmail.cf, don't forget to 
regenerate sendmail.cf -- something like the following:

cd /etc/mail; cp sendmail.cf sendmail.cf.todaysdate; m4  sendmail.mc  
sendmail.cf

should work, followed by /etc/init.d/sendmail restart

Mojo


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Re: [asterisk-users] merge gsm files

[Mojo with Horan Company, LLC]

Tzafrir Cohen wrote:
 On Tue, Dec 11, 2007 at 11:26:21PM +0800, Rilawich Ango wrote:
   
 Hi,
  How can I merge 2 gsm files into a single file?  I have tried to use
 soxmix as below but failed.
 soxmix 1.gsm 2.gsm 1-2.gsm
 

 The GNU coreutils are shipped with a special[1] tool for this task: 

 cat 1.gsm 2.gsm 1-2.gsm

 [1] Honestly, it is special. I bet the BSD cat can't do that. And I bet
 that the busybox one can't do that either.

   
I thought he meant mix so they play simultaneously, not concatenate them 
together.

To clarify Tzafrir's comment a little, I think that 'cat' will only work 
for sound files if all involved are headerless.  I'm not 100% sure which 
ones are and which ones aren't headerless, but generally, this shouldn't 
work for a standard wave file.   An image showing WHY it won't work can 
be found at http://ccrma.stanford.edu/CCRMA/Courses/422/projects/WaveFormat/

Keep in mind that some files that were once headerless, like .au, can 
now be found with six 32-bit words prepended as a header, sort of a 
'new' .au format.  These would NOT be concatenatable either, unless 
those first words were stripped from the second file.  Further, the 
first file would need its third 32-bit word modified to either 
0x or the actual *new* size of the data in bytes.

Sorry, but I can't help with mixing them rather than concatenating them :)

Mojo


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[asterisk-users] Realtime: Should I say or should I go (now) ?

[Olivier]

Hi,

I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should consider using Asterisk Realtime and a
database to manage phones registrations.

Stories in Dev mailing list say Realtime is mis-used or should be improved.
So, what's the bottom line ?
Can I consider anything I can do with .conf files can be done with a
combination of .conf files and Realtime.

Regards
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Re: [asterisk-users] ip phone suggestion for Asia?

[Mindaugas Kezys]

Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. 

We use them (SPA942) in our company. Everybody's happy.


Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky
Sent: Thursday, December 20, 2007 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ip phone suggestion for Asia?

Hi:
   i am surveying ip phones for our company. we will use them with asterisk.
   we have office in taiwan, hong kong,singapore and china.
   cisco and polycom are too expensive for us.
   we try several china brand ip phones. they are all cheap and
some of them have good quality. but most of them won't offer future firmware
support, which we think it's important for ip phones.
   searching in the mail list, we found aastra is good, but they don't sale to
asia. grandstream looks good also.there are many grandstream users in the list,
can someone share any good or bad experience about grandstream today?
   if there are other good choice, please tell us!!
   thanks a lot for your help!!

Regards,
tbskyd

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Re: [asterisk-users] turn off auto-seek extention - force usetimeout

[Justin Killen]

Hey, that works great!

Thanks!


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Mojo with Horan  Company, LLC
 Sent: Wednesday, December 19, 2007 5:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] turn off auto-seek extention - force
 usetimeout
 
 So I'm guessing this is what you're doing:


--
 
 [ids]
 exten = s,1,playback(enter your id number)
 exten = s,2,WaitExten(10)
 exten = s,3,Goto(1)
 
 exten = 4768,1,blahblahblah
 exten = 4790,1,blahblahblah
 exten = 4732,1,blahblahblah
 
 exten = i,1,playback(error)
 exten = i,2,goto(s,1)


--
 
 So, maybe place the phones in a context that waits for a four-digit id
 _before_ matching it to the context you were initially trying:


--
 
 [getid]
 exten = s,1,playback(enter your id number)
 exten = s,2,WaitExten(10)
 exten = s,3,Goto(1)
 
 exten = _4XXX,1,goto(ids,${exten},1)
 
 [ids]
 exten = 4768,1,blahblahblah
 exten = 4790,1,blahblahblah
 exten = 4732,1,blahblahblah
 
 exten = i,1,playback(error)
 exten = i,2,goto(getid,s,1)
 


--
 
 
 Untested: I wonder if one entered an extension that didn't exist, say
 4555, when we tried to Goto(ids, 4555, 1) would we get directed to
 extension i in the extensions context or would the call be dropped
 completely?
 
 Moj
 
 Justin Killen wrote:
 
  I have an application where a call-in user is prompted to enter an
  identification number for schedule information. That id number is
  setup as an extension, and if that extension doesn't exist, it tells
  them that they are not scheduled, then loops back to ask for the id
  number again. My problem is that asterisk pre-emptively goes to the
i
  extension (invalid) too early depending on available extensions. For
  example, if I put in id number 4768, and there is only 4790 and
4732,
  it will push to the invalid extension on the 6, then the not
  scheduled playback message (a cepstral command) gets cancelled out
  from the DTMF push of the 8. So, if I put in 4768, I get prompted to
  enter an id number. What I would like to do is turn off this
feature,
  so that the number input does not get evaluated until after the
  timeout (preferably configurable from the extensions.conf file).
 
  Thanks in advance
 
  -Justin
 
 

 
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Re: [asterisk-users] merge gsm files

[Steve Edwards]

On Thu, 20 Dec 2007, Mojo with Horan  Company, LLC wrote:

 Tzafrir Cohen wrote:

 On Tue, Dec 11, 2007 at 11:26:21PM +0800, Rilawich Ango wrote:

 Hi,
  How can I merge 2 gsm files into a single file?  I have tried to use

 The GNU coreutils are shipped with a special[1] tool for this task:

 cat 1.gsm 2.gsm 1-2.gsm

 [1] Honestly, it is special. I bet the BSD cat can't do that. And I bet
 that the busybox one can't do that either.

 To clarify Tzafrir's comment a little, I think that 'cat' will only work
 for sound files if all involved are headerless.  I'm not 100% sure which

Nope. Took me off guard as well, so I had to try it. cat on my CentOS 
box can append gsm files and they play seamlessly.

Thanks in advance,

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

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[asterisk-users] MeetMeConference

[bilal ghayyad]

Hi All;

Is there any limitation on the number of users for
MeetMe Conference? In other words, how many parties
can join to the room and become a member of the room?
Is there any limitation?

Regards
Bilal


  

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[asterisk-users] H323 and Gatekeeper

[bilal ghayyad]

Hi List;

In the h323.conf file, the parameter gatekeeper is
used to let asterisk work as h323 gatekeeper listening
at port 1719 by setting gatekeeper=DISCOVER or it is
used to let asterisk search for the gatekeeper to talk
with it and receive calls from it? But if just to let
asterisk talk with it, then what asterisk will talk
with it other than receiving calls from it?

Any help?
Regards
Bilal


  

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Re: [asterisk-users] MeetMeConference

[Dean Collins]

as far as I know it's unlimited and only tied to the capacity of a
single machines processing power.
Of course then all you need to do is tie multiple machines to the same
room in a daisy chain and expand from there.
Someone on the list a few months ago gave the example of chaining
together 5 servers this way - sorry i cant remember who it was.

Cheers,
Dean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal
ghayyad
Sent: Thursday, 20 December 2007 2:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MeetMeConference

Hi All;

Is there any limitation on the number of users for MeetMe Conference? In
other words, how many parties can join to the room and become a member
of the room?
Is there any limitation?

Regards
Bilal


 


Be a better friend, newshound, and
know-it-all with Yahoo! Mobile.  Try it now.
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ 


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Re: [asterisk-users] ip phone suggestion for Asia?

[Daniel Cole]

We currently also use the Linksys SPA942 and SPA963 IP phones. They are very 
nice phones, and very easy to manage.

Cheers,

Daniel Cole  (CCNA)
Technical Support

Ph: 1800 424 683
Fax: 03 5221 7659
e: [EMAIL PROTECTED]
w: hugonet.com.au

---

The information transmitted is the property of HugoNet and is intended only for 
the person or entity to which it is addressed and may contain confidential 
and/or privileged material. Statements and opinions expressed in this e-mail 
may not represent those of the company. Any review, retransmission, 
dissemination and other use of, or taking of any action in reliance upon, this 
information by persons or entities other than the intended recipient is 
prohibited. If you received this in error, please contact the sender 
immediately and delete the material from any computer.

 P Please consider the environment before you print this e-mail or any 
attachments.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mindaugas Kezys
Sent: Friday, 21 December 2007 6:03 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ip phone suggestion for Asia?

Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive.

We use them (SPA942) in our company. Everybody's happy.


Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky
Sent: Thursday, December 20, 2007 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ip phone suggestion for Asia?

Hi:
   i am surveying ip phones for our company. we will use them with asterisk.
   we have office in taiwan, hong kong,singapore and china.
   cisco and polycom are too expensive for us.
   we try several china brand ip phones. they are all cheap and some of them 
have good quality. but most of them won't offer future firmware support, which 
we think it's important for ip phones.
   searching in the mail list, we found aastra is good, but they don't sale to 
asia. grandstream looks good also.there are many grandstream users in the list, 
can someone share any good or bad experience about grandstream today?
   if there are other good choice, please tell us!!
   thanks a lot for your help!!

Regards,
tbskyd

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[asterisk-users] Asterisk and Chan_h323: all calls are not going

[bilal ghayyad]

Hi All;

I established h323 trunk using chan_h323 (between
asterisk and softswitch, already i did this using sip
and succeed, but now in h323 and i am facing a
problem).

The call reached to the softswitch, but it always
dropped when it send for destination, and we tried to
let softswitch to send to different gateways, but
always drop.

I tried with h245Tunneling=yes and with no, also with
fastStart=yes and with no, and in softswitch I tried
with media and signaling proxy and i tried with only
signaling, ... all of attempts was giving in the end a
dropping in the call. The same settings on the
softswitch for same routes, it worked using endpoints
other than asterisk (cisco, avaya, ip phones,
softphones).

Any one has idea?
My current asterisk version is 1.4.7

Any advise?

Regards
Bilal


  

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[asterisk-users] Telco MWI Detection on TDM400 Interface?

[Jim Duda]

I use a TDM400 card to interface with my telco.  I used asterisk voice 
mail.  However, if I'm on the telco line while another call comes in, 
obviously it cannot go to asterisk voice mail but instead bounces to the 
Telco voice mail.

Is there any means by which I can get asterisk detect the Message 
Waiting Indicator (MWI) from the telco?  Is there some application or 
variable which can be used to identify an active MWI from the Telco?

Thanks,

Jim


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Re: [asterisk-users] Telco MWI Detection on TDM400 Interface?

[Brian J. Murrell]

On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
 I use a TDM400 card to interface with my telco.  I used asterisk voice 
 mail.  However, if I'm on the telco line while another call comes in, 
 obviously it cannot go to asterisk voice mail but instead bounces to the 
 Telco voice mail.
 
 Is there any means by which I can get asterisk detect the Message 
 Waiting Indicator (MWI) from the telco?  Is there some application or 
 variable which can be used to identify an active MWI from the Telco?

Heh.  And go one step further and go retrieve the message from the Telco
and put it into an Asterisk voice mail message.  :-)

b.



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Re: [asterisk-users] ip phone suggestion for Asia?

[Joe]

I always forget about Linksys/Sipura phones. Yes they are very nice. I know
there can be some contractual issues when you deal with Linksys/Cisco, but
they are well constructed and I haven't heard anything bad.

I do think you'd get better support with Snom/Grandstream, but this is
coming from my experiences with Cisco more then Linksys.



On Dec 20, 2007 3:24 PM, Daniel Cole [EMAIL PROTECTED] wrote:

 We currently also use the Linksys SPA942 and SPA963 IP phones. They are
 very nice phones, and very easy to manage.

 Cheers,

 Daniel Cole  (CCNA)
 Technical Support

 Ph: 1800 424 683
 Fax: 03 5221 7659
 e: [EMAIL PROTECTED]
 w: hugonet.com.au


 ---

 The information transmitted is the property of HugoNet and is intended
 only for the person or entity to which it is addressed and may contain
 confidential and/or privileged material. Statements and opinions expressed
 in this e-mail may not represent those of the company. Any review,
 retransmission, dissemination and other use of, or taking of any action in
 reliance upon, this information by persons or entities other than the
 intended recipient is prohibited. If you received this in error, please
 contact the sender immediately and delete the material from any computer.

  P Please consider the environment before you print this e-mail or any
 attachments.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] On Behalf Of Mindaugas Kezys
 Sent: Friday, 21 December 2007 6:03 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] ip phone suggestion for Asia?

 Do not forget to evaluate Linksys SPA phones. Best I tried and not
 expensive.

 We use them (SPA942) in our company. Everybody's happy.


 Regards,
 Mindaugas Kezys
 http://www.kolmisoft.com
 MOR - Advanced Billing for Asterisk PBX


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] On Behalf Of d tbsky
 Sent: Thursday, December 20, 2007 6:34 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] ip phone suggestion for Asia?

 Hi:
   i am surveying ip phones for our company. we will use them with
 asterisk.
   we have office in taiwan, hong kong,singapore and china.
   cisco and polycom are too expensive for us.
   we try several china brand ip phones. they are all cheap and some of
 them have good quality. but most of them won't offer future firmware
 support, which we think it's important for ip phones.
   searching in the mail list, we found aastra is good, but they don't sale
 to asia. grandstream looks good also.there are many grandstream users in
 the list, can someone share any good or bad experience about grandstream
 today?
   if there are other good choice, please tell us!!
   thanks a lot for your help!!

 Regards,
 tbskyd

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Re: [asterisk-users] ip phone suggestion for Asia?

[John Novack]

Daniel Cole wrote:
 We currently also use the Linksys SPA942 and SPA963 IP phones. They are very 
 nice phones, and very easy to manage.
   
Both the 842 and 942 do not have lit displays and are hard to read.
Is the 963 any better?
Why anyone EVER built a LCD without backlighting is beyond me.

John Novack


 Cheers,

 Daniel Cole  (CCNA)
 Technical Support

 Ph: 1800 424 683
 Fax: 03 5221 7659
 e: [EMAIL PROTECTED]
 w: hugonet.com.au

 ---

 The information transmitted is the property of HugoNet and is intended only 
 for the person or entity to which it is addressed and may contain 
 confidential and/or privileged material. Statements and opinions expressed in 
 this e-mail may not represent those of the company. Any review, 
 retransmission, dissemination and other use of, or taking of any action in 
 reliance upon, this information by persons or entities other than the 
 intended recipient is prohibited. If you received this in error, please 
 contact the sender immediately and delete the material from any computer.

  P Please consider the environment before you print this e-mail or any 
 attachments.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mindaugas Kezys
 Sent: Friday, 21 December 2007 6:03 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] ip phone suggestion for Asia?

 Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive.

 We use them (SPA942) in our company. Everybody's happy.


 Regards,
 Mindaugas Kezys
 http://www.kolmisoft.com
 MOR - Advanced Billing for Asterisk PBX


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky
 Sent: Thursday, December 20, 2007 6:34 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] ip phone suggestion for Asia?

 Hi:
i am surveying ip phones for our company. we will use them with asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco and polycom are too expensive for us.
we try several china brand ip phones. they are all cheap and some of them 
 have good quality. but most of them won't offer future firmware support, 
 which we think it's important for ip phones.
searching in the mail list, we found aastra is good, but they don't sale 
 to asia. grandstream looks good also.there are many grandstream users in the 
 list, can someone share any good or bad experience about grandstream today?
if there are other good choice, please tell us!!
thanks a lot for your help!!

 Regards,
 tbskyd

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Re: [asterisk-users] MeetMeConference

[dave cantera]

also, you want to think about transcoding... if you have different 
technologies, the system load for transcoding would increase...

dean, cool, I didn't know you could hang a few * boxes together with 
meetme...
daveC


Dean Collins wrote:
 as far as I know it's unlimited and only tied to the capacity of a
 single machines processing power.
 Of course then all you need to do is tie multiple machines to the same
 room in a daisy chain and expand from there.
 Someone on the list a few months ago gave the example of chaining
 together 5 servers this way - sorry i cant remember who it was.

 Cheers,
 Dean

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of bilal
 ghayyad
 Sent: Thursday, 20 December 2007 2:44 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] MeetMeConference

 Hi All;

 Is there any limitation on the number of users for MeetMe Conference? In
 other words, how many parties can join to the room and become a member
 of the room?
 Is there any limitation?

 Regards
 Bilal


  
 
 
 Be a better friend, newshound, and
 know-it-all with Yahoo! Mobile.  Try it now.
 http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ 


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-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894




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Re: [asterisk-users] Telco MWI Detection on TDM400 Interface?

[Russell Bryant]

Brian J. Murrell wrote:
 On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
 Is there any means by which I can get asterisk detect the Message 
 Waiting Indicator (MWI) from the telco?  Is there some application or 
 variable which can be used to identify an active MWI from the Telco?

This functionality was just added to Asterisk trunk.  See the mwimonitor 
option.

 Heh.  And go one step further and go retrieve the message from the Telco
 and put it into an Asterisk voice mail message.  :-)

But no, it can't do this part yet.  Though I bet you could do it with some
really creative dialplan plus the use of the mwimonitornotify option to generate
an outbound call.  :)

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] ip phone suggestion for Asia?

[Daniel Cole]

Hi,

We currently use the 942 and 962. These both have backlight displays. The 962 
is obviously the model with the nice color screen.
These phones are very easy to manage. We configure all phone remotely via TFTP, 
and they phones are set to pull their config periodically. Very easy for making 
changes off-site :)


Cheers,


Daniel Cole  (CCNA)
Technical Support

Ph: 1800 424 683
Fax: 03 5221 7659
e: [EMAIL PROTECTED]
w: hugonet.com.au

---

The information transmitted is the property of HugoNet and is intended only for 
the person or entity to which it is addressed and may contain confidential 
and/or privileged material. Statements and opinions expressed in this e-mail 
may not represent those of the company. Any review, retransmission, 
dissemination and other use of, or taking of any action in reliance upon, this 
information by persons or entities other than the intended recipient is 
prohibited. If you received this in error, please contact the sender 
immediately and delete the material from any computer.

 P Please consider the environment before you print this e-mail or any 
attachments.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, 21 December 2007 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ip phone suggestion for Asia?



Daniel Cole wrote:
 We currently also use the Linksys SPA942 and SPA963 IP phones. They are very 
 nice phones, and very easy to manage.

Both the 842 and 942 do not have lit displays and are hard to read.
Is the 963 any better?
Why anyone EVER built a LCD without backlighting is beyond me.

John Novack


 Cheers,

 Daniel Cole  (CCNA)
 Technical Support

 Ph: 1800 424 683
 Fax: 03 5221 7659
 e: [EMAIL PROTECTED]
 w: hugonet.com.au

 --
 -

 The information transmitted is the property of HugoNet and is intended only 
 for the person or entity to which it is addressed and may contain 
 confidential and/or privileged material. Statements and opinions expressed in 
 this e-mail may not represent those of the company. Any review, 
 retransmission, dissemination and other use of, or taking of any action in 
 reliance upon, this information by persons or entities other than the 
 intended recipient is prohibited. If you received this in error, please 
 contact the sender immediately and delete the material from any computer.

  P Please consider the environment before you print this e-mail or any 
 attachments.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Mindaugas Kezys
 Sent: Friday, 21 December 2007 6:03 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] ip phone suggestion for Asia?

 Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive.

 We use them (SPA942) in our company. Everybody's happy.


 Regards,
 Mindaugas Kezys
 http://www.kolmisoft.com
 MOR - Advanced Billing for Asterisk PBX


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky
 Sent: Thursday, December 20, 2007 6:34 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] ip phone suggestion for Asia?

 Hi:
i am surveying ip phones for our company. we will use them with asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco and polycom are too expensive for us.
we try several china brand ip phones. they are all cheap and some of them 
 have good quality. but most of them won't offer future firmware support, 
 which we think it's important for ip phones.
searching in the mail list, we found aastra is good, but they don't sale 
 to asia. grandstream looks good also.there are many grandstream users in the 
 list, can someone share any good or bad experience about grandstream today?
if there are other good choice, please tell us!!
thanks a lot for your help!!

 Regards,
 tbskyd

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Re: [asterisk-users] ip phone suggestion for Asia?

[Jared Smith]

On Thu, 2007-12-20 at 17:17 -0500, John Novack wrote:
 Both the 842 and 942 do not have lit displays and are hard to read.

Actually, the 942 *does* have a backlit LCD.

 Is the 963 any better?

I'm assuming that's a typo and they really meant the 962.  The 962 has a
bright color backlit LCD.

 Why anyone EVER built a LCD without backlighting is beyond me.

I tend to agree.

---
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] put fxo channel before E1 channel?

[Jared Smith]

On Thu, 2007-12-20 at 23:39 +0800, d tbsky wrote:
 hi:
  my system has one 4-port fxo card and one 2-port E1 card.
  for some reason, i like to place fxo as channel 1-4, and E1 use the
 rest channels (5-66).

This will only work if you load the kernel driver for the fxo card
before the kernel driver for the E1 card.  To see which order they've
come up in, you can check the /proc/zaptel directory.  You should see a
file in that directory for each span, and if you look at the contents of
each file, you'll see which channels are in that span.

--
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Cisco 7961 new firmware stops reading configuration files

[Preston Edwards]

Chad,

I had the same problem when upgrading to some of the newer firmware. The newer 
firmware gets even pickier (if that's even possible) about the config files. Go 
the phone's webpage and look at the debug log. It will show you where it's not 
parsing correctly. I'm not in front of my phone now so I can't look, but I 
remember it getting upset about networkLocale or userLocale or something of 
that nature, so I just removed that section of the XML code and it loaded fine.

Good luck,
Preston Edwards

- Original Message 
From: Chad Osmond [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 20, 2007 9:32:43 AM
Subject: [asterisk-users] Cisco 7961 new firmware stops reading configuration 
files


Hello,
 
I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have
recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front
of the phone and also to hopefully resolve some issues with the phones
not registering after a long period.

Once we upgraded the phones now display Error Verifying Config Info
 in
the Status messages and will not process the configuration file.

To make a change on the phone I have to downgrade to 8.2.2R4 and change
the configuration, and then upgrade to 8.3.2R1, which is a bit of a
pain.

The tftp logs indicate that the phones is getting the correct
SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be
the issue.

The Wiki pages for 79x1 indicate that it's a known issue, has anyone
managed to get past the issue?
I tried logging a call with Cisco TAC, but they're giving the We don't
support SIP on anything other then CME...

Thanks,

Chad


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[asterisk-users] Failed Call Debugging?

[Atlanticnynex]

My PHP script is using AMI's Originate command to make two-way calls.
The originate connects the first leg of the calls, plays a file to the
first called party, and then uses Dial() from the dialplan to dial the
other leg of the call.

I'm noticing that only about 30% of the calls make it through successfully.
In looking at Asterisk's CDR's, I have noticed a lot of NO ANSWER and
even more FAILED call dispositions. It seems that there is a patter of
NO ANSWER being followed by a FAILED immediately after. Although, some
records break this (either vice-versa or a single record of one of them).

My problem is that I am under pretty high volume (say 500+ calls) a day,
and I don't have any way of figuring out what is causing this. Whenever
I test my script manually, I don't have the problem.

Somethings that come to mind are: Lack of Server Resources, VoIP
Provider issues, and Invalid Phone #'s.

I doubt my VoIP provider, Asterlink, is not able to handle my volume
(usually no more than 6 or 8 calls simultaneously). My server load
averages are far below alarm, and I have more than 50MB of free RAM when
6 calls are going through. I have checked for invalid phone numbers, and
that is not the case either.

Here's a couple things I noticed were showing up in the
/var/log/asterisk/messages quite frequently:

WARNING[2220] cdr.c: CDR already initialized on '**Unknown**'  #this
shows up a lot!

and

WARNING[2897] cdr.c: Cause not handled
 WARNING[2266] chan_sip.c: Remote host can't match request BYE to call
'[EMAIL PROTECTED]'. Giving up.



Any suggestions on how to better troubleshoot this?

Thanks!
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Re: [asterisk-users] Telco MWI Detection on TDM400 Interface?

[Mojo with Horan Company, LLC]

I'm reminded of the (Pagoo??) call waiting voicemail applet a while back.

Let me sum up how one might do this:  I'm using vitelity for an 800# 
DID... it's $0.50/mo plus time used, which is around $0.02/min maybe.  
get your asterisk box registered to your itsp so that when your 800# DID 
is dialed you enter the same dialplan your pstn callers fall into.   
voila, your home now has an 800#. 

Anyway, get your phone company to 'forward on busy' TO your 800#.  So 
when your home phone is busy, the caller will instead be connected to 
your asterisk via the internet and not even notice the difference (well, 
if they were long distance.  Local callers might wonder why your 
voicemail prompts sound like they're long distance though ;) )

I'm not being commercial about vitelity, just an example of what 
services they provide that could achieve this.

Mojo

Jim Duda wrote:
 I use a TDM400 card to interface with my telco.  I used asterisk voice 
 mail.  However, if I'm on the telco line while another call comes in, 
 obviously it cannot go to asterisk voice mail but instead bounces to the 
 Telco voice mail.

 Is there any means by which I can get asterisk detect the Message 
 Waiting Indicator (MWI) from the telco?  Is there some application or 
 variable which can be used to identify an active MWI from the Telco?

 Thanks,

 Jim


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Re: [asterisk-users] turn off auto-seek extention - force usetimeout

[Mojo with Horan Company, LLC]

Justin Killen wrote:
 Hey, that works great!

 Thanks!

   
 -Original Message-
 So, maybe place the phones in a context that waits for a four-digit id
 _before_ matching it to the context you were initially trying:
 
Excellent :) No problem at all!

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Re: [asterisk-users] put fxo channel before E1 channel?

[d tbsky]

hi jsmith:

   that explains everything. i didn't aware the module load sequence would
cause big difference. is there any document about this i am missing?
now the system is working as expected. i m glad that i asked and you
answered.
thanks a lot for your quick reply and help!!

Regards,
tbskyd


2007/12/21, Jared Smith [EMAIL PROTECTED]:
 On Thu, 2007-12-20 at 23:39 +0800, d tbsky wrote:
  hi:
   my system has one 4-port fxo card and one 2-port E1 card.
   for some reason, i like to place fxo as channel 1-4, and E1 use the
  rest channels (5-66).

 This will only work if you load the kernel driver for the fxo card
 before the kernel driver for the E1 card.  To see which order they've
 come up in, you can check the /proc/zaptel directory.  You should see a
 file in that directory for each span, and if you look at the contents of
 each file, you'll see which channels are in that span.

 --
 Jared Smith
 Community Relations Manager
 Digium, Inc.


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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Dovid B]

Dave,
I agree with you. I think it would be smarter to go to a new format how ever 
one issues that a lot of people seem to have is when the syntax is changed. 
This is why I suggested both. Maybe there can be a month (or maybe even two) 
long discussion between the users and dev list for A) Current formatting B) 
formatting for the future and we can have both say for the next two major 
releases (as opposed to 1 now) and then move over. Wouldn't this make more 
people happy ?


- Original Message - 
From: dave cantera [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, December 20, 2007 6:33 AM
Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!


 dovid...
 while this seems like a good idea to have both sip show channels and
 show channels sip having two, three or even four ways to do the same
 thing would confuse/cripple the learning curve... * would turn into a
 microsoft mentality where there are dozens of ways to
 configure/reconfigure some of their products...  word, for example, can
 be configured with or without the tool bars and then you can configure
 hot-keys...  in fact, you can configure some products so that someone
 who learns it with a hacked config, could not possibly use the original
 stock config...  sorry to go on about this but it is one of my hot
 buttons...
 daveC

 Dovid B wrote:
 - Original Message - 
 From: Steve Edwards [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, December 19, 2007 5:43 AM
 Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's 
 old!



 On Sat, 15 Dec 2007, Johansson Olle E wrote:


 I wonder if there are any major obstacles for upgrading.

 How about the change from a bad command line interface to a really bad
 command line interface?

 I mean, Seriously? (in a Grey's Anatomy kind of way...)

 The old syntax was inconsistent -- show manager command vs sip show
 channels and just plain bad -- for example sip reload should have 
 been
 reload sip.

 The new syntax continues down the noun-verb path instead of correcting
 itself and using verb-noun like most other applications (MySQL, GDB,
 Oracle, etc.)

 Then, just to make it worse, now I have to learn which commands somebody
 (arbitrarily) decided are core and which are not -- for what benefit?
 Certainly doesn't make MY job easier!

 Approach the command line like a noob. I want Asterisk to show me
 something so I'll start the command line with show. I'm not quite sure
 what I'm doing, so I'll press TAB to see what I can show. Oh, 
 channel
 looks like what I want. Hmm, too much. Maybe I should have qualified 
 what
 kind of channel I'm looking for BEFORE the word channel.

 Here's a suggestion -- stop thinking like a parser and start thinking 
 like
 a person :)

 Which makes more sense (at least in English)?

  1) show black dogs -- show sip channels
  2) black show dogs -- sip show channels
  3) dogs black show -- channels sip show
  4) show dogs black -- show channels sip
  5) black dogs show -- sip channels show
  6) dogs show black -- channels show sip

 Is it too late to fix this for 1.6?

 Thanks in advance,


 I think as many people have pointed out they are used to a lot of 
 commands
 out there so changing it yet again would make more people unhappy. But 
 maybe
 asterisk can have both. Why not sip show channels for the old timers and
 show channels sip or show sip channels for the n00b's. Why shouldn't
 asterisk have both options ?



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 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

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 856.380.0894




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Re: [asterisk-users] AsteriskNOW release date???

[Raúl Gómez C.]

Thank you very much Jared, this are good news!

Thanks again...

On Dec 19, 2007 7:03 PM, Jared Smith [EMAIL PROTECTED] wrote:

 I just got off the phone with the software product manager who is over
 AsteriskNOW, and have it on good authority that it will be released in
 early January.  There were a few last-minute things that weren't quite
 ready, but as of today things appear to be back on track and almost
 ready for release.

 ---
 Jared Smith
 Community Relations Manager
 Digium, Inc.


-- 
Nacho
Linux Counter #156439
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Re: [asterisk-users] Telco MWI Detection on TDM400 Interface?

[Jim Duda]

Thanks Russell, that's what I'm looking for.

Any idea when this will become part an official asterisk release?

Jim

Russell Bryant wrote:
 Brian J. Murrell wrote:
 On Thu, 2007-12-20 at 16:29 -0500, Jim Duda wrote:
 Is there any means by which I can get asterisk detect the Message 
 Waiting Indicator (MWI) from the telco?  Is there some application or 
 variable which can be used to identify an active MWI from the Telco?
 
 This functionality was just added to Asterisk trunk.  See the mwimonitor 
 option.
 
 Heh.  And go one step further and go retrieve the message from the Telco
 and put it into an Asterisk voice mail message.  :-)
 
 But no, it can't do this part yet.  Though I bet you could do it with some
 really creative dialplan plus the use of the mwimonitornotify option to 
 generate
 an outbound call.  :)
 


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Re: [asterisk-users] Realtime: Should I say or should I go (now) ?

[Benjamin Jacob]

I am using Realtime in virtually all my projects. So far, I haven't had 
any major issues. It saves a lot of headache for profile/dialplan 
updates, at least for me!
So I say, GO!

- Ben

Olivier wrote:

 Hi,

 I'm working on a 500 seats Asterisk project.
 I'm wondering whether or not I should consider using Asterisk Realtime 
 and a database to manage phones registrations.

 Stories in Dev mailing list say Realtime is mis-used or should be 
 improved.
 So, what's the bottom line ?
 Can I consider anything I can do with .conf files can be done with a 
 combination of .conf files and Realtime.

 Regards



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Re: [asterisk-users] Upgrade to Asterisk 1.4 - testing

[Johansson Olle E]

20 dec 2007 kl. 01.43 skrev Dovid B:

 snip
 Our problem is that very few in the community test beta releases
 or development code. I want to send a big thank you to all that do,
 you are very important in this process. And for those of you who
 want to join, go to www.asterisk.org and find instructions on how
 to download development code for testing. Join the whoever
 tests this stuff group today :-)
 /snip

 Olle I would love to test but I do not know what I am looking for. I  
 would
 say that I have a fairly good knowledge of Asterisk  however I am  
 not the
 best at tracing the root problems of issues. I have no problem of  
 loading
 the bleeding edge version on a spate box, loading my current configs  
 on it
 and seeing where it goes down. Maybe some info on what to look for  
 when
 there are issues would help.

Dovid,
For people that wants to help the process, there's always time and
a large attention span from the development team. Join the #asterisk-dev
channel on IRC freenode.net and you'll find a weird enivronment (many
jokes among friends) but also a lot of people that can help you get  
going,
give you ideas for testing and respond to your ideas. There's usually
a lot of real-time activity there (US time, not on my mornings here in
Sweden at GMT+1), but it might slow down now for Xmas.

Welcome!

/O

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