[asterisk-users] Asterisk takes incoming call before extension was submitted

[m . tanzer]

I have a Problem with incoming ISDN calls in Austria, I use zaptel and asterisk 
bristuffed from Debian/Etch.

- If someone outside is dialing the phonenumber and the extension on an ISDN 
phone, asterisk catches the call and puts into in the s extension before the 
extension was submittet.
- The same Number dialed from a mobilephone works as expected, asterisk 
recieves the extension. As well with stored numbers in ISDN phones.

From Diskussions in german Forums it turns out, that Austria does not follow 
the DSS1 Standard (what ever this is...) and there is a timing Problem with 
the zapata ISDN configuration.

Does anyone know, how I can increase the timout value for the ISDN implses, so 
asterisk waits for the extension on the ISDN channel? 

Regards, martin

My zapata.conf:

[trunkgroups]

[channels]

language=de
pridialplan=local
prilocaldialplan=local
nationalprefix = 00
internationalprefix = 000
; trust user provided callerid (clip no screening)?
pritrustusercid = yes
; hidecallerid=no
callerid=asreceived

switchtype = euroisdn
signalling = bri_cpe
pridialplan = local
prilocaldialplan = local
echocancel = yes
echocancelwhenbridged=no
echotraining=no
usecallerid = yes
overlapdial = yes
immediate = no
group = 1
context = isdn
channel = 1-2

switchtype = euroisdn
signalling = bri_cpe
pridialplan = local
prilocaldialplan = local
echocancel = yes
echocancelwhenbridged=no
echotraining=no
usecallerid = yes
overlapdial = yes
immediate = no
group = 1
context = isdn
channel = 4-5

echotraining = yes
rxgain = 0.8
txgain = 0.8
signalling = fxo_ks
context = default
channel = 7

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Re: [asterisk-users] Queue Penalties not working properly

[Syed Nasruddin]

I am using asterisk 1.4.18. I cant at this stage upgrade to any latest
version. Linear strategy for queues is not in asterisk 1.4.18. I have to
use ringall instead.

Is it possible Disabling call-waiting for my agents only?? While other
sip users have call waiting functionality.

regards 
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robin
Rodriguez
Sent: Tuesday, August 05, 2008 11:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue Penalties not working properly

Syed Nasruddin wrote:
 Hi,

 Actully the way I want the penalties functionality to behave it is not
 doing it accordingly. I am right now using ringall. Set penalty 1 for
 one agent and 2 for secnd agent. All the calls come in and go to first
 agent#1 having penalty one. But the second call also go to agent#1 and
 start waiting for it to be free rather it should have gone to penalty
 two agent#2

 I have added call-limit=1 for bot sip accounts. And started the
 services. Still find the status wrong.

 nasr

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Atis
 Lezdins
 Sent: Tuesday, August 05, 2008 7:03 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Queue Penalties not working properly

 On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED]
 wrote:
   
  Cannot i use ringall strategy with penalties???

 Will rrmemory will fullfil my requirement??
 

 rrmemory isn't ringall, it won't ring all members. But yes - you can
 use ringall with penalties.

   
 My requirements:


 1. 10 Call Center Agents.

 2.   All the calls coming in will ALWAYS be routed to specific 5
 
 agents,
   
 firstly.

 3. IF ALL the first 5 agents are busy then ONLY then the call will be
 routed to next 5 Agents.


 Moreover why my queue status shows my agent as NOT IN USE while in
 
 fact
   
 it is busy answering the call??
 

 What you are seeing is caused by status NOT IN USE. You have to set
 call-limit in sip.conf for all your phones, to any value, so that
 device states work correctly, and queue can know that those phones are
 busy. Now you probably can see in CLI that queue is sending second
 call to first agent(s).

 Regards,
 Atis



   
 Thanks

 Syed nasr


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steve
 Totaro
 Sent: Tuesday, August 05, 2008 5:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Queue Penalties not working properly

 On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez
 [EMAIL PROTECTED] wrote:
 
 Syed Nasruddin wrote:
   
 Hi,

 I am using Asterisk 1.4.18. I am implementing Penalties for my
 
 agents.
 
 What is happening: two agents configuired one agent with penalty 1
 
 and
 
 the other with penalty 2. All the calls must go first to Agent 1
and
 if his line is busy then only then agent 2 will get the call.
 
 However
   
 my queues are not behaving in this manner. I have impmemnted
ringall
 strategy. Now when first call comes it ends up with agent 1, when
 secnd call comes it continue wait in queue and doesn't go to agent
2
 and when agent one is free it goes to this agent.

 I have set penalties in queue.conf. I have monitered my queue and
 witnessed that my agent1 status shows Not In Use and Agent 2 also
 
 same
 
 status is this the reason behind this. I have copied my queue show
 results below.please help . how do I change this stauts problem

 callcenter*CLI queue show

 myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s
 holdtime), W:0, C:2, A:0, SL:0.0% within 0s

 Members:

 SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was
 
 2233
   
 secs ago)

 SIP/1000 with penalty 2 (Not in use) has taken no calls yet

 No Callers

 Syed nasr


 
 You need to use the linear queue strategy, it is in 1.6 or there
is
   
 a
 
 backport to 1.4

 --
 Robin Rodriguez
 VoIP/Telecom Engineer
 Atlantic.net
 1-800-211-9496

   
 Robin, round robin

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Very carefully reread the descriptions on penalties and queue strategies

on voip-info.org, the first time I 

Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 13

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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[asterisk-users] does astcanary really work?

[Pavel Jezek]

A week ago, I tried give realtime priority to asterisk proces using -p 
switch,
asterisk was running inside astcanary,
but yestarday asterisk probably starts eating all cpu and lock any 
access to computer, only ping was possible,
so, anybody have experience, that ascanary process does really work to 
lower process priority in case of overloading?
PJ

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Re: [asterisk-users] Grandstream RS-232 config (slightly off-topic)

[Vieri]

--- On Tue, 8/5/08, Atis Lezdins [EMAIL PROTECTED] wrote:

 Have you tried powering it on, while holding reset button?

Yes, several times. I contacted Grandstream's helpdesk and they told me to keep 
the reset button pressed while I plug the power off and back on again. I even 
tried keeping it pressed for 5 minutes and the device kept rebooting but the 
factory defaults were never restored.

 Additionally you can try to leave it for week powered off
 and hope
 that there's some old battery keeping up settings.

I'll try that... :-(

 Are you sure that there isn't some enable admin
 mode command in
 telnet? It should allow you everything that's available
 from web.

I wish there were but I haven't found anything. I even tried enable admin 
mode (just in case this was my lucky day) but it doesn't compute...

I suspect that the built-in HTTP server crashes (which would be a GS bug) as 
soon as the device boots maybe because of a problem with the configuration. 
Since I can't reset to defaults I tried to setup a default config file 
(cfgMAC file) on my LAN HTTP provisioning server (whose URL I had previously 
setup in the faulty device). When the GS device boots I can see from my Apache 
log that the cfgMAC file is being fetched OK. However, the device's 
configuration is left unchanged. So it loads the cfgMAC but it doesn't commit 
it/store it or something. Something's definitely wrong there.

Curiously, I checked two other GXW4008 on my LAN (both are working fine). 
However, if I connect to both via telnet, I can see that one of them has a 
reset to factory defaults command and the other doesn't (just like the faulty 
device). As you can see here the settings on both devices are identical:

ATA #13:
Grandstream GXW-4008  V1.3A Command Shell Copyright 2006-2008
Supported commands:
config  -- Configure the device
status  -- Show device status
upgrade -- Upgrade the device
reboot  -- Reboot the device
reset   -- Factory reset
help-- Show this help text
exit-- Exit this command shell
Software Versions:
Main -- 1.0.0.86
Boot -- 1.0.0.7
Core -- 1.0.0.21
Base -- 1.0.0.60

ATA #2:
Grandstream GXW-4008  V1.3A Command Shell Copyright 2006-2008
Supported commands:
config  -- Configure the device
status  -- Show device status
upgrade -- Upgrade the device
reboot  -- Reboot the device
help-- Show this help text
exit-- Exit this command shell
Software Versions:
Main -- 1.0.0.86
Boot -- 1.0.0.7
Core -- 1.0.0.21
Base -- 1.0.0.60

Really odd (both systems have customized configs).

Anyway, thanks for your feedback. I hope GS will come up with a new firmware 
with a complete list of telnet commands.

Vieri



  

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[asterisk-users] shared mysql connection in dialplan

[Rizwan Hisham]

hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received and a new mysql
connection is made using the MYSQL cmd in dialplan. i want to use a single
mysql connection for every incoming call.

my idea of doing it is like this, i want to get a mysql connection in a
global variable, just to share the connection with different incoming calls.
Im not sure if this can be done. I am going to try doing it somehow,
meanwhile i want your suggestions about how i can share a mysql connection
with different calls in a dialplan.

I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
connectivity.

Thanx in advance

-- 
Best Regards
Rizwan Hisham
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Re: [asterisk-users] Asterisk to Avaya

[Steve Davies]

Good question, I'll check.

Regards,
Steve

2008/8/6 Tom Lynn [EMAIL PROTECTED]:
 Steve, what kind of Avaya system is this?  They make several.

 On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:

 Hi,

 Sorry this is so long, but I am reasonably desparate.

 I am having real fun with hooking an Avaya system to Asterisk using
 ISDN30. I have the ISDN signalling all sorted one way, and can pass
 calls from the real world (ie. the telco and asterisk) TO the avaya
 box, and it accepts that and sets up the call perfectly.

[...]

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[asterisk-users] About the features.conf of it's transfer

[larry]

HI
   This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
 Thanks!
  LARRY
[general]
parkext = 700
parkpos = 701-702

context = parkedcalls

[featuremap]
atxfer = *2

[applicationmap]
set(DYNAMIC_FEATURES=tranf)

tranf = *2,peer,waitexten(10|m)


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[asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing

[Patrick]

Hi,

My apologies for the OT. My googling came up empty and hopefully there 
are some members in the community that could give me a hint how to solve 
this issue:

Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. 
The downgrade process started off good. The 7961 got it's IP address via 
DHCP, found it's SEPmac.cnf.xml file and started to upgrade the 
phone with the 8.0.4 firmware. All was well until it finally rebooted. 
Now it get's an IP from the DHCP server and says upgrading. Nothing 
else. It just seems to hang (monitored it for more than an hour).

Anyone have an idea how I can fix this?

Thanks and regards,
Patrick

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[asterisk-users] asterisk realtime user deletion

[Nhadie]

Hi All,

Would just like to know if anyone has encountered this:

i a user is currently registered using SPA 941, i then tried deleting 
the user in the realtime db. then i tried to make a call from the SPA i 
can still make calls even though user has been deleted.

i tried the same thing this time using an x-lite, i'm registered on 
x-lite, i deleted user in the db, x-lite cannot make calls, whcih should 
be the proper case.

i tried same thing with zoiper, i got the same result as the x-lite.

i have the rtcachefriends set to no, but why my SPA can still make calls 
when the xlite/zoiper cannot? TIA

Regards
Ron

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[asterisk-users] Transcoding

[Guilherme Loch Waltrick Góes]

I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM
format. I have these same prompts in another server with Asterisk 1.4.18, on
this server the prompts sound pretty nice, but on the first one they sound
pretty choppy. Was there any changes on the transcoding code between this 2
versions ? Any hints ?
Best Regards,

-- 
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
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Re: [asterisk-users] Asterisk to Avaya

[Steve Davies]

I am told it is an IP Office 400 series.

I have not been on site physically which does not help.

Regards,
Steve

2008/8/6 Tom Lynn [EMAIL PROTECTED]:
 Steve, what kind of Avaya system is this?  They make several.

 On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:

 Hi,

 Sorry this is so long, but I am reasonably desparate.

 I am having real fun with hooking an Avaya system to Asterisk using
 ISDN30. I have the ISDN signalling all sorted one way, and can pass
 calls from the real world (ie. the telco and asterisk) TO the avaya
 box, and it accepts that and sets up the call perfectly.

 The problem is that the Avaya box is signalling outbound calls using
 an odd method, which smacks of an analogue system with ISDN30 bolted
 on for a bit of a laugh.

 They send a q931 SETUP message. This contains the correct callerID,
 but only the first 1 to 4 of the dialled number's digits - The
 remainder of the number is I believe passed through using DTMF!!! From
 the look of it they intentionally do not send an IE 161 Sending
 Complete with the SETUP, so that the far end continues to listen for
 these DTMF tones, until it resolves to a legal number.

 My questions for some ISDN expert out there...

 Part 1)  I need to receive the number in the SETUP, which I guess will
 be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,
 and check the dialplan to see if it is a locally terminated number.
 Once I am 100% sure it is not local, I can then dial the collected
 number through the Telco ISDN channel. Make sense? I think I can
 probably handle that. The problem is that I do not know whether I have
 received all digits from the Avaya at that point, which leads to...

 Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a
 difference) without sending the IE 161 call complete? I thought that
  Dial(Zap/G1||D(${INITIAL}))
 might send the initial digits using DTMF, and then leave the channel
 open so that more DTMF could follow over the now bridged channel. In
 fact I get an immediate failure as if the far end thinks I have
 finished dialling. Can I assume that libpri does not currently support
 this method of dialling? If not, how might it be added? I can hack the
 code, I just need suggestions of where to look and how it might sanely
 be added :)

 Part 3) It is possible that the Avaya is not using DTMF at-all, and
 that it will send more bits of the called-party number using the
 D-Channel as you would expect, but Asterisk does not seem to be
 waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone
 know the Avaya systems well enough to suggest how it might be working?

 Many many thanks for any feedback.

 Regards,
 Steve

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[asterisk-users] problem with iaxmodem!

[nboumediene]

Hello,

I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk
and I work on Redhat.
I installed the two hylafax and iaxmodem.
My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0)

device /dev/ttyIAX0
owner uucp:uucp
mode 660
port 4570 #each line should have it's own port number!
refresh 300
server 127.0.0.1
peername IAXmodem #this is the local extension number in FreePBX (create it!)
secret 12345 #password for the extension
cidname Fax1
cidnumber 
codec ulaw

I added this two lines in /etc/inittab

IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0
mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0

Then I tried to configure hylafax whith the command faxsetup

I meet a problem when I want to add a modem with the command faxaddmodem
but I can't, I have this response:

Serial port that modem is connected to []? ttyIAX0
/dev/ttyIAX0 is not a terminal device.

In fact in /dev I don't find ttyIAX0

I added the line
 /usr/sbin/faxgetty -D /dev/ttyIAX0  in the file
/etc/rc.d//rc.local

and I tried the command faxgetty -D /dev/ttyIAX0

but nothing!!!

I tried the command /usr/local/bin/iaxmodem ttyIAX0 
I have the following response:

[EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0
[2008-08-05 17:39:27] Modem started
[2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0'
[2008-08-05 17:39:27] Setting owner = 'uucp:uucp'
[2008-08-05 17:39:27] Setting mode = '660'
[2008-08-05 17:39:27] Setting port = 4570
[2008-08-05 17:39:27] Setting refresh = 300
[2008-08-05 17:39:27] Setting server = '127.0.0.1'
[2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local
extension number in FreePBX (create '
[2008-08-05 17:39:27] Setting secret = '12345 #password for the extension'
[2008-08-05 17:39:27] Setting cidname = 'Fax1'
[2008-08-05 17:39:27] Setting cidnumber = ''
[2008-08-05 17:39:27] Setting codec = ulaw
[2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17
[2008-08-05 17:39:27] Removed old /dev/ttyIAX0
[2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link
[2008-08-05 17:39:27] Registration failed.


I don't unerstand why iaxmodem can't register .
If someone has an idea, he is welcome!!
Thank you



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Re: [asterisk-users] Transcoding

[Darren Sessions]

I would make absolutely sure you've got your linux distro's version of  
libgsm installed. I can't really speak to the difference between those  
two versions of Asterisk without looking at a change-log, but I highly  
doubt a serious modification to the gsm code took place between sub- 
versions.


Hope this helps,

 - Darren


_

[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_



On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote:

I have a server with Asterisk 1.4.21.1 and some prompts recorded in  
GSM format. I have these same prompts in another server with  
Asterisk 1.4.18, on this server the prompts sound pretty nice, but  
on the first one they sound pretty choppy. Was there any changes on  
the transcoding code between this 2 versions ? Any hints ?


Best Regards,

--
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre: 
http://www.asteriskexperts.com.br
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Re: [asterisk-users] problem with iaxmodem!

[Atis Lezdins]

On Wed, Aug 6, 2008 at 4:05 PM,  [EMAIL PROTECTED] wrote:

 Hello,

 I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk
 and I work on Redhat.
 I installed the two hylafax and iaxmodem.
 My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0)

 device /dev/ttyIAX0
 owner uucp:uucp
 mode 660
 port 4570 #each line should have it's own port number!
 refresh 300
 server 127.0.0.1
 peername IAXmodem #this is the local extension number in FreePBX (create it!)
 secret 12345 #password for the extension
 cidname Fax1
 cidnumber 
 codec ulaw

 I added this two lines in /etc/inittab

 IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0
 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0

 Then I tried to configure hylafax whith the command faxsetup

 I meet a problem when I want to add a modem with the command faxaddmodem
 but I can't, I have this response:

 Serial port that modem is connected to []? ttyIAX0
 /dev/ttyIAX0 is not a terminal device.

 In fact in /dev I don't find ttyIAX0

 I added the line
  /usr/sbin/faxgetty -D /dev/ttyIAX0  in the file
 /etc/rc.d//rc.local

 and I tried the command faxgetty -D /dev/ttyIAX0

 but nothing!!!

 I tried the command /usr/local/bin/iaxmodem ttyIAX0 
 I have the following response:

 [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0
 [2008-08-05 17:39:27] Modem started
 [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0'
 [2008-08-05 17:39:27] Setting owner = 'uucp:uucp'
 [2008-08-05 17:39:27] Setting mode = '660'
 [2008-08-05 17:39:27] Setting port = 4570
 [2008-08-05 17:39:27] Setting refresh = 300
 [2008-08-05 17:39:27] Setting server = '127.0.0.1'
 [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local
 extension number in FreePBX (create '
 [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension'
 [2008-08-05 17:39:27] Setting cidname = 'Fax1'
 [2008-08-05 17:39:27] Setting cidnumber = ''
 [2008-08-05 17:39:27] Setting codec = ulaw
 [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17
 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0
 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link
 [2008-08-05 17:39:27] Registration failed.


 I don't unerstand why iaxmodem can't register .
 If someone has an idea, he is welcome!!
 Thank you


What's your iax.conf? For me modem configuration looks like this:

[iaxmodem5]
type=friend
host=dynamic
secret=x
context=fax
permit=127.0.0.1
allow=all

P.S. after editing inittab, you also have to execute

# kill -HUP 1

So that init process re-reads configuration.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing

[Matt Gibson]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Sent: Wednesday, August 06, 2008 7:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 -
8.0.4SRS2 failing

Hi,

My apologies for the OT. My googling came up empty and hopefully there 
are some members in the community that could give me a hint how to solve 
this issue:

Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. 
The downgrade process started off good. The 7961 got it's IP address via 
DHCP, found it's SEPmac.cnf.xml file and started to upgrade the 
phone with the 8.0.4 firmware. All was well until it finally rebooted. 
Now it get's an IP from the DHCP server and says upgrading. Nothing 
else. It just seems to hang (monitored it for more than an hour).

Anyone have an idea how I can fix this?

Thanks and regards,
Patrick







Did you change your SEPXXX when you upgraded to 8.3.3? You may have to
revert those changes. Check the debug log on the phones web interface to see
if it's choking on a particular line in the cfg.

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com


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Re: [asterisk-users] Transcoding

[Guilherme Loch Waltrick Góes]

I'm using OpenSUSE 10.3, the funny thing is: if the softphone is using GSM
the sounds is perfect, if I use Alaw as the softphone CODEC the sounds is
pretty bad. The softphone is in the same LAN as the Asterisk server, so I
don't think it's a bandwidth issue.
Best Regards,


On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions [EMAIL PROTECTED]wrote:

 I would make absolutely sure you've got your linux distro's version of
 libgsm installed. I can't really speak to the difference between those two
 versions of Asterisk without looking at a change-log, but I highly doubt a
 serious modification to the gsm code took place between sub-versions.
 Hope this helps,

  - Darren


 _

 [EMAIL PROTECTED]
 http://www.darrensessions.com
 http://www.linkedin.com/in/dsessions
 _



 On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote:

 I have a server with Asterisk 1.4.21.1 and some prompts recorded in GSM
 format. I have these same prompts in another server with Asterisk 1.4.18, on
 this server the prompts sound pretty nice, but on the first one they sound
 pretty choppy. Was there any changes on the transcoding code between this 2
 versions ? Any hints ?
 Best Regards,

 --
 Guilherme Loch Góes

 Visite nossa loja virtual: http://www.shopvoip.com.br

 Notícias e Fórum sobre VoIP com software livre:
 http://www.asteriskexperts.com.br
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-- 
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
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Re: [asterisk-users] Transcoding

[Darren Sessions]

I am a **BIG, BIG** fan of OpenSUSE.

:)

Use yast under 'Software Management' and do a search for 'gsm'.

Make sure gsmlib and gsmlib-devel are *both* installed. Then scroll  
down and make sure that libgsm and libgsm-devel are *both* installed.


After that, you'll have to recompile Asterisk.

See if that does anything for you.

 - Darren



_

[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_



On Aug 6, 2008, at 8:48 AM, Guilherme Loch Waltrick Góes wrote:

I'm using OpenSUSE 10.3, the funny thing is: if the softphone is  
using GSM the sounds is perfect, if I use Alaw as the softphone  
CODEC the sounds is pretty bad. The softphone is in the same LAN as  
the Asterisk server, so I don't think it's a bandwidth issue.


Best Regards,


On Wed, Aug 6, 2008 at 10:13 AM, Darren Sessions  
[EMAIL PROTECTED] wrote:
I would make absolutely sure you've got your linux distro's version  
of libgsm installed. I can't really speak to the difference between  
those two versions of Asterisk without looking at a change-log, but  
I highly doubt a serious modification to the gsm code took place  
between sub-versions.


Hope this helps,

 - Darren


_

[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_



On Aug 6, 2008, at 7:02 AM, Guilherme Loch Waltrick Góes wrote:

I have a server with Asterisk 1.4.21.1 and some prompts recorded in  
GSM format. I have these same prompts in another server with  
Asterisk 1.4.18, on this server the prompts sound pretty nice, but  
on the first one they sound pretty choppy. Was there any changes on  
the transcoding code between this 2 versions ? Any hints ?


Best Regards,

--
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre: 
http://www.asteriskexperts.com.br
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--
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre: 
http://www.asteriskexperts.com.br
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Re: [asterisk-users] vars in Macros called by DIAL with option M

[Jay R. Ashworth]

On Tue, Aug 05, 2008 at 02:13:15PM -0500, Tilghman Lesher wrote:
 above the original post is very confusing.  Please stop doing this.
 The format of this post is in reverse, to demonstrate why posting a reply
 
 option is only in trunk.  So no, it would not help him out.
 Yes, it works the same way, by using the U() option to Dial.  However, this
 Question 2:
 
 prior to the origination of the slave channel.
 channel.  No inheritance is possible, because the master channel originated
 channel, so any values set in the slave channel will not affect the master
 Apple.  The variable is only set in the slave channel, not in the master
 Question 1:

And the award for Best Illustration of a Point goes to...

Tilghman Lesher!

Mr Lesher has been nominated for this award 4 times; this is his first
win.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Transcoding

[Mark Michelson]

Guilherme Loch Waltrick Góes wrote:
 I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some 
 prompts recorded in GSM format. I have these same prompts in another 
 server with Asterisk 1.4.18, on this server the prompts sound pretty 
 nice, but on the first one they sound pretty choppy. Was there any 
 changes on the transcoding code between this 2 versions ? Any hints ?
 
 Best Regards, 
 
 -- 
 Guilherme Loch Góes
 
 Visite nossa loja virtual: http://www.shopvoip.com.br
 
 Notícias e Fórum sobre VoIP com software livre: 
 http://www.asteriskexperts.com.br

One important difference between the servers may be the compiler used. We have 
heard reports that using GCC 4.2 or later with optimizations on causes choppy 
audio when using GSM.

Solutions to this include either downgrading your compiler to GCC 4.1 or 
earlier, or selecting DONT_OPTIMIZE in menuselect under compiler options and 
then recompiling Asterisk. I also believe that you can set the optimization 
level for compilation to -O2 in Makefile.rules and have no choppy audio, but I 
cannot confirm this.

Of course, if this server isn't running GCC 4.2, then you can ignore everything 
I've said so far :)

Mark Michelson

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Re: [asterisk-users] Transcoding

[Tilghman Lesher]

On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
 I would make absolutely sure you've got your linux distro's version of
 libgsm installed. I can't really speak to the difference between those
 two versions of Asterisk without looking at a change-log, but I highly
 doubt a serious modification to the gsm code took place between sub-
 versions.

There was one slight change, which will only make a difference if you're
using gcc 4.2 or above.  The change was to fix a new optimization in gcc
4.2 that caused some inline assembly to be incorrectly built, which
corrupted sound.

-- 
Tilghman

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Re: [asterisk-users] problem with iaxmodem!

[Nadjia Boumédiène]

My iax.conf looks like this:

[iaxmodem]
type=friend
host=127.0.0.1
secret=x
context=fax-out
permit=127.0.0.1
disallow=all
allow=ulaw

after editing inittab I reload it by running: /sbin/init q

I also reboot the system with shutdown -r now and I had the following
message:
init: Id mo respawning too fast: disabled for 5 minutes.

I don't know what it signifies! 

Regards,

Nadjia Boumediene, 
Legos
[EMAIL PROTECTED]
Work phone:+ 33172292995


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Atis Lezdins
Envoyé : mercredi 6 août 2008 15:46
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] problem with iaxmodem!

On Wed, Aug 6, 2008 at 4:05 PM,  [EMAIL PROTECTED] wrote:

 Hello,

 I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk
 and I work on Redhat.
 I installed the two hylafax and iaxmodem.
 My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0)

 device /dev/ttyIAX0
 owner uucp:uucp
 mode 660
 port 4570 #each line should have it's own port number!
 refresh 300
 server 127.0.0.1
 peername IAXmodem #this is the local extension number in FreePBX (create
it!)
 secret 12345 #password for the extension
 cidname Fax1
 cidnumber 
 codec ulaw

 I added this two lines in /etc/inittab

 IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0
 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0

 Then I tried to configure hylafax whith the command faxsetup

 I meet a problem when I want to add a modem with the command faxaddmodem
 but I can't, I have this response:

 Serial port that modem is connected to []? ttyIAX0
 /dev/ttyIAX0 is not a terminal device.

 In fact in /dev I don't find ttyIAX0

 I added the line
  /usr/sbin/faxgetty -D /dev/ttyIAX0  in the file
 /etc/rc.d//rc.local

 and I tried the command faxgetty -D /dev/ttyIAX0

 but nothing!!!

 I tried the command /usr/local/bin/iaxmodem ttyIAX0 
 I have the following response:

 [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0
 [2008-08-05 17:39:27] Modem started
 [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0'
 [2008-08-05 17:39:27] Setting owner = 'uucp:uucp'
 [2008-08-05 17:39:27] Setting mode = '660'
 [2008-08-05 17:39:27] Setting port = 4570
 [2008-08-05 17:39:27] Setting refresh = 300
 [2008-08-05 17:39:27] Setting server = '127.0.0.1'
 [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local
 extension number in FreePBX (create '
 [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension'
 [2008-08-05 17:39:27] Setting cidname = 'Fax1'
 [2008-08-05 17:39:27] Setting cidnumber = ''
 [2008-08-05 17:39:27] Setting codec = ulaw
 [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17
 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0
 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link
 [2008-08-05 17:39:27] Registration failed.


 I don't unerstand why iaxmodem can't register .
 If someone has an idea, he is welcome!!
 Thank you


What's your iax.conf? For me modem configuration looks like this:

[iaxmodem5]
type=friend
host=dynamic
secret=x
context=fax
permit=127.0.0.1
allow=all

P.S. after editing inittab, you also have to execute

# kill -HUP 1

So that init process re-reads configuration.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] Action on login

[Stefan Gofferje]

Hi,

is there meanwhile the possibility for some actions besides dialling in *?
Namely, I would like that if a remote IAX or SIP user logs in AND there
are new messages, they automatically get a call and be connected to the
voicemail. The only method I know by now is make a context in the
dialplan, checking if the user has logged in and then initiate the call.
And of course firing a callfile to every x minutes to that context for
each remote user. That does not scale very well. It would be much nicer
to have some kind of login / logout action parameter in sip.conf or so.

--Stefan


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Re: [asterisk-users] Transcoding

[Darren Sessions]

I have used virtually all versions of Asterisk 1.0+ (literally, either  
in production or testing) with OpenSUSE 10+ and 11 on AMD and Intel  
and haven't had any issues with gcc optimizations with regards to  
audio sounding choppy. This scenario for me has always been the gsm  
libs.



_

[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_



On Aug 6, 2008, at 9:16 AM, Mark Michelson wrote:


Guilherme Loch Waltrick Góes wrote:

I have a server with Asterisk 1.4.21.1 http://1.4.21.1 and some
prompts recorded in GSM format. I have these same prompts in another
server with Asterisk 1.4.18, on this server the prompts sound pretty
nice, but on the first one they sound pretty choppy. Was there any
changes on the transcoding code between this 2 versions ? Any hints ?

Best Regards,

--
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br


One important difference between the servers may be the compiler  
used. We have
heard reports that using GCC 4.2 or later with optimizations on  
causes choppy

audio when using GSM.

Solutions to this include either downgrading your compiler to GCC  
4.1 or
earlier, or selecting DONT_OPTIMIZE in menuselect under compiler  
options and
then recompiling Asterisk. I also believe that you can set the  
optimization
level for compilation to -O2 in Makefile.rules and have no choppy  
audio, but I

cannot confirm this.

Of course, if this server isn't running GCC 4.2, then you can ignore  
everything

I've said so far :)

Mark Michelson

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Re: [asterisk-users] problem with iaxmodem!

[Karsten Wemheuer]

Hi,

Am Mittwoch, den 06.08.2008, 17:24 +0200 schrieb Nadjia Boumédiène:
 My iax.conf looks like this:
 
 [iaxmodem]
 type=friend
 host=127.0.0.1
 secret=x
 context=fax-out
 permit=127.0.0.1
 disallow=all
 allow=ulaw
 
 after editing inittab I reload it by running: /sbin/init q
 
 I also reboot the system with shutdown -r now and I had the following
 message:
 init: Id mo respawning too fast: disabled for 5 minutes.

this message indicates, that the service (identified through id mo)
died short after it's start. So the init-process starts it again and
again. Cause this happens to fast, it disables the restarting of the
process.

Having a look at Your inittab in Your first post, I would suggest to
remove the -D switch from the line with faxgetty. This command line
switch instructs the faxgetty to detach from terminal. In this case the
init process looses contact to the process and tries to restart it.

HTH,
Karsten



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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 14

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 - 8.0.4SRS2 failing

[Patrick]

Hi Matt,

Thank you for your suggestion. Comment inline.

Matt Gibson wrote:
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Patrick
 Sent: Wednesday, August 06, 2008 7:34 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] OT: Cisco 7961 SIP downgrade from 8.3.3 -
 8.0.4SRS2 failing
 
 Hi,
 
 My apologies for the OT. My googling came up empty and hopefully there 
 are some members in the community that could give me a hint how to solve 
 this issue:
 
 Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. 
 The downgrade process started off good. The 7961 got it's IP address via 
 DHCP, found it's SEPmac.cnf.xml file and started to upgrade the 
 phone with the 8.0.4 firmware. All was well until it finally rebooted. 
 Now it get's an IP from the DHCP server and says upgrading. Nothing 
 else. It just seems to hang (monitored it for more than an hour).
 
 Anyone have an idea how I can fix this?
 
 Thanks and regards,
 Patrick 
 
 
 Did you change your SEPXXX when you upgraded to 8.3.3? You may have to
 revert those changes. Check the debug log on the phones web interface to see
 if it's choking on a particular line in the cfg.

Just to make sure, it's a downgrade from 8.3.3 to 8.0.4. The only line 
changed in the SEPxxx file is:

from
loadInformationSIP41.8-3-3S/loadInformation

to
loadInformationSIP41.8-0-4SRS2/loadInformation

Nothing else changed. When I power the phone down and up again, the only 
thing it does is going into the upgrade screen, getting an IP from the 
DHCP server and then say Upgrading. In the tftpserver logs I can see 
that it does not even pick up the SEPxxx file.

I haven't checked the phone's web interface. Frankly I wasn't aware that 
it was active during the upgrade process. I did try telnet and ssh but 
both were unresponsive. Will try the web interface when I'm near the 
phone again.

Thanks,
Patrick

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Re: [asterisk-users] in-call start monitoring

[Bill Michaelson]

I suppose, too. So see below. I also verified that the dial command is 
using Ww (which I had to fudge), but still, no monitoring.

Anything else I can check?

pig*CLI feature show
Builtin Feature Default Current
--- --- ---
Pickup *8 *8
Blind Transfer # #
Attended Transfer
One Touch Monitor *1
Disconnect Call * **
Park Call

Dynamic Feature Default Current
--- --- ---
(none)

Call parking

Parking extension : 70
Parking context : parkedcalls
Parked call extensions: 71-79

 From: Paul Hales [EMAIL PROTECTED]

 I suppose the bit to check is the features ('show features') and then 
 try to record a call (*1) and see what the terminal says...


 Bill Michaelson wrote:
   
  My client needs call recording features and would like to initiate the 
  process in-call (typically *1).  I'm installing Asterisk 1.4.x and 
  FreePBX 2.4+.  I'm using Polycom phones.  I can't make it work.  Would 
  somebody please give a checklist of items for me to compare my list 
  against - in the hope I've overlooked something?


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Re: [asterisk-users] Asterisk to Avaya

[Steve Davies]

2008/8/5 Steve Davies [EMAIL PROTECTED]:
 Hi,

 Sorry this is so long, but I am reasonably desparate.

 I am having real fun with hooking an Avaya system to Asterisk using
 ISDN30. I have the ISDN signalling all sorted one way, and can pass
 calls from the real world (ie. the telco and asterisk) TO the avaya
 box, and it accepts that and sets up the call perfectly.

 The problem is that the Avaya box is signalling outbound calls using
 an odd method, which smacks of an analogue system with ISDN30 bolted
 on for a bit of a laugh.

[...]

Okay, I think I am progressing in terms of my understanding.

Firstly, I had missed out overlapdial=yes from the inbound PRI_NET
channel from the Avaya. I have not been able to check, but that should
allow Asterisk to collect the remaining digits until it finds a match
in the dialplan. This begs the following questions:

If an inbound overlapdial uniquely matches:
  exten = _X.,1,NoOp()

Then I assume it will match after any 6 digits have been received, and
drop into the dialplan. Given that this is a Zap channel, how do I
receive any subsequent digits if they are dialled? Are they converted
to inband DTMF??? I cannot find any useful documentation on what
overlapdial=yes really does - Pointers welcome.

Also, what if the overlap dialled number will never be unique, so I
need to trap both:
  exten = _X.,1,NoOp()
and
  exten = 01234567890,1,NoOp()

Will overlapdial ever start executing one of those 2 patterns if I
dial 01234567890 and nothing else? I appreciate that I could probably
do the following instead - perhaps I've answered my own question?
  exten = 01234567890,1,NoOp()
  exten = _!.,1,Goto(passthru,${EXTEN},1)

Then, importantly, how do I overlap dial outbound using Zaptel? The
Dial() command is designed to send a number and wait for a response.
There is no opportunity for further input AFAIK. Does enabling
overlapdial=yes mean that I can Dial() and it will not assume the
number is complete?

Perhaps Asterisk simply cannot do this? It is a pretty horrible requirement!

Regards,
Steve

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[asterisk-users] Regarding fmtp parameters.

[SiM]

Hello All,
   I'am doing a video call between two Video Phones, and i see
that Asterisk is stripping the fmtp parameters for the h263 video line in
SDP.
For example a line similar to the below is stripped,

 a=fmtp:xx CIF=4;QCIF=2;F=1;K=1

Asterisk is configured NOT to be present in the Media path (My version :
Asterisk 1.4.19.1 ).
I have the following enabled in my sip.conf.

canreinvite=yes
directrtpsetup=yes

From what i have read on the internet, i feel fmtp parameters are not
supported by Asterisk for Video.
I also find that video_caps branch has a fix for this problem, please can
someone share more information
about this and where i can find it ?

I do not want those fmtp lines to be stripped. Suggestions to change the
Asterisk config files, to achieve this are also welcome.

Thank you.


Best regards,
Simith
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Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 15

[dimitri . osler]

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a 
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al 
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], 
altrimenti vi risponderò al mio rientro.

Dimitri Osler

I will be on vacation until Tuesday 19th of August with limited access to voice 
and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 
0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on 
my return.

Dimitri Osler





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Re: [asterisk-users] Transcoding

[Tzafrir Cohen]

On Wed, Aug 06, 2008 at 10:15:00AM -0500, Tilghman Lesher wrote:
 On Wednesday 06 August 2008 08:13:09 Darren Sessions wrote:
  I would make absolutely sure you've got your linux distro's version of
  libgsm installed. I can't really speak to the difference between those
  two versions of Asterisk without looking at a change-log, but I highly
  doubt a serious modification to the gsm code took place between sub-
  versions.
 
 There was one slight change, which will only make a difference if you're
 using gcc 4.2 or above.  The change was to fix a new optimization in gcc
 4.2 that caused some inline assembly to be incorrectly built, which
 corrupted sound.

Also note that the issue was triggered by uisng -O3. I know that at
least most Debian packages are built with -O2 .

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Max amount of concurrent calls on and iax trunk

[Rosli Sukri]

hi,
wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes
connected via iax trunk. have u guys ever stress tested the trunks i.e how
many concurrent calls can a trunk handle and whether codec has any effect on
it.
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[asterisk-users] Strange beep during calls

[Felippe Silvestre]

Hi all,
 
Our users are complaining about beeps that happen in the middle of some
calls. They are similar to the sound heard you are in a call and press
any button in your phone. Please find bellow some examples of these
beeps(the recordings are in Portuguese, but the beeps are easy to
identify):
 
http://www.katizak.locaweb.com.br/asterisk/beep.mp3
http://www.katizak.locaweb.com.br/asterisk/beep.mp3 
http://www.katizak.locaweb.com.br/asterisk/beep2.mp3
http://www.katizak.locaweb.com.br/asterisk/beep3.mp3
http://www.katizak.locaweb.com.br/asterisk/beep4.mp3
 
We are sure that our users are not pressing any button in the softphones
during the conversations.
Do you guys are able to identify where these beeps are coming from?
Maybe an * functionality that we need to turn off... We are using
Asterisk 1.4.21.2.
 
Thanks.
 
Felippe Silvestre
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Re: [asterisk-users] Strange beep during calls

[Ruddy Gbaguidi]

maybe you are using the L option in Dial app to limit the conversation time.
Check those channel variables (just a wild guess)
*LIMIT_PLAYAUDIO_CALLER
**LIMIT_PLAYAUDIO_CALLEE
**LIMIT_TIMEOUT_FILE
**LIMIT_CONNECT_FILE
**LIMIT_WARNING_FILE

*
Felippe Silvestre wrote:
 Hi all,
  
 Our users are complaining about beeps that happen in the middle of 
 some calls. They are similar to the sound heard you are in a call and 
 press any button in your phone. Please find bellow some examples of 
 these beeps(the recordings are in Portuguese, but the beeps are easy 
 to identify):
  
 http://www.katizak.locaweb.com.br/asterisk/beep.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3
  
 We are sure that our users are not pressing any button in the 
 softphones during the conversations.
 Do you guys are able to identify where these beeps are coming from? 
 Maybe an * functionality that we need to turn off... We are using 
 Asterisk 1.4.21.2.
  
 Thanks.
  
 Felippe Silvestre
 

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 Internal Virus Database is out of date.
 Checked by AVG. 
 Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 
 7:42 PM
   


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Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

[Robin Rodriguez]

Rosli Sukri wrote:
 hi,
 wanted to ask if anybody has experienced setting up two asterisk 1.2 
 boxes connected via iax trunk. have u guys ever stress tested the 
 trunks i.e how many concurrent calls can a trunk handle and whether 
 codec has any effect on it.
 

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What are the hardware specs of the boxes, and what is the speed of the 
connection between them?



-- 
Robin Rodriguez
VoIP/Telecom Engineer
Atlantic.net
1-800-211-9496


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[asterisk-users] asterisk realtime user deletion

[Nhadie]

Hi All,

Would just like to know if anyone has encountered this:

i a user is currently registered using SPA 941, i then tried deleting
the user in the realtime db. then i tried to make a call from the SPA i
can still make calls even though user has been deleted.

i tried the same thing this time using an x-lite, i'm registered on
x-lite, i deleted user in the db, x-lite cannot make calls, whcih should
be the proper case.

i tried same thing with zoiper, i got the same result as the x-lite.

i have the rtcachefriends set to no, but why my SPA can still make calls
when the xlite/zoiper cannot? TIA

Regards
Ron

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[asterisk-users] Digium B410P: problematic Bri connection between * and a legacy Philips PBX

[Daniele Visaggio]

Hi all,

my goal is to connect my trixbox server (CentOS 5.2 - kernel
2.6.18-53.1.4.el5 - * 1.4.20-1) with a legacy Philips PBX with 4 bri
links provided from Digium B410P.

For this reason I set all the 4 ports of Digium's card in NT mode
(Philips can not do this). Then i opportunely
edited /etc/misdn-init.conf and /etc/asterisk/misdn.conf. In fact, when
I run the command misdn shows stacks in * CLI, I can see all ports in
NT (PTP) mode. Once i connect the wires from Philips PBX to the Digium's
card, L2 and L1 go immediately up and i can do any calls from * to
Philips and viceversa...only for 10 minutes! Then from Philips console I
can see that B-channels of every port become not usable. From that
moment any calls form Philips to * fail but: 

1) I can still place calls from * to Philips
2) misdn show stacks output does not underline any problem (all 4 ports
appear up with L2/L1).

In your opinion, how can i fix this? why this problem after 10 minutes?

Thank you - Daniele

p.s.: I already attempt to change some options
in /etc/asterisk/misdn.conf file. I tried to put
incoming_early_audio=yes (explanation: Rarely used. If turned on, sends
Tone Indications on TE Port for Incoming isdn channel. Normally the
telcos send that informaton. By default is 'no'). It didn't work...


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Re: [asterisk-users] asterisk realtime user deletion

[Tilghman Lesher]

On Wednesday 06 August 2008 15:07:03 Nhadie wrote:
 Would just like to know if anyone has encountered this:

You sent the exact same email this morning at 7:47 a.m.  If nobody has
responded, it's because nobody has ever seen that before.  Duplicating
the message tells everybody on the list that you have low regard for their
time and bandwidth.

-- 
Tilghman

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[asterisk-users] intercom/paging with grandstream gxp2000

[Fidel Garcia]

Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom.  I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with variables and flags.

 

Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.

 

Extensions.conf:

 

exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)

exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)

exten=s,2,SIPAddHeader(Call-Info: answer-after=0)

exten=s,3,Dial(${ARG2},20)

exten=s,4,Goto(s-${DIALSTATUS},1)

exten=s-NOANSWER,1,Voicemail(${ARG1},u)

exten=s-NOANSWER,2,Goto(default,s,1)

exten=s-BUSY,1,Voicemail(${ARG1},b)

exten=s-BUSY,2,Goto(default,s,1)

exten=_s-.,1,Goto(s-NOANSWER,1)

exten=a,1,VoicemailMain(${ARG1})

 

GXP2000 configuration:

Under Account1 I checked options:

 

Allow Auto Answer by Call-Info:   No  Yes  

 

Turn off speaker on 
remote disconnect:   No  Yes

 

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: [EMAIL PROTECTED] 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

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Re: [asterisk-users] intercom/paging with grandstream gxp2000

[Fidel Garcia]

I am sorry, this is the actual extensions.conf:

exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)

exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)

exten=s,3,SIPAddHeader(Call-Info: answer-after=0)

exten=s,4,Dial(${ARG2},20)

exten=s,5,Goto(s-${DIALSTATUS},1)

exten=s-NOANSWER,1,Voicemail(${ARG1},u)

exten=s-NOANSWER,2,Goto(default,s,1)

exten=s-BUSY,1,Voicemail(${ARG1},b)

exten=s-BUSY,2,Goto(default,s,1)

exten=_s-.,1,Goto(s-NOANSWER,1)

exten=a,1,VoicemailMain(${ARG1})

 

As you can see here Goto and SIPAddHeader are 2 and 3. In the  prior email I
had both lines under 2.

 

Fidel Garcia

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fidel Garcia
Sent: Wednesday, August 06, 2008 5:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] intercom/paging with grandstream gxp2000

 

Guys I have been reading for days on how to get this to work with asterisk
and for some reason every time I call the call goes to intercom.  I know I
must be doing something wrong with the way I am adding the steps to my call;
I am not familiar with variables and flags.

 

Here is my configuration:
Digium Asterisk AA50 with Granstream GXP2000 using the latest firmware.

 

Extensions.conf:

 

exten=s,1,SIPAddHeader(Alert-Info: http://127.0.0.1\;info=Family)

exten=s,2,GotoIf($[${SIP_HEADER(Call-Info)}=answer-after=0]?2:3)

exten=s,2,SIPAddHeader(Call-Info: answer-after=0)

exten=s,3,Dial(${ARG2},20)

exten=s,4,Goto(s-${DIALSTATUS},1)

exten=s-NOANSWER,1,Voicemail(${ARG1},u)

exten=s-NOANSWER,2,Goto(default,s,1)

exten=s-BUSY,1,Voicemail(${ARG1},b)

exten=s-BUSY,2,Goto(default,s,1)

exten=_s-.,1,Goto(s-NOANSWER,1)

exten=a,1,VoicemailMain(${ARG1})

 

GXP2000 configuration:

Under Account1 I checked options:

 

Allow Auto Answer by Call-Info:   No  Yes  

 

Turn off speaker on 
remote disconnect:   No  Yes

 

 

 

Fidel Garcia

System Engineer

 

sysTeam.

7205 NW 19th Street, Suite 302
Miami, Florida 33126

Email: [EMAIL PROTECTED] 

Tel: (305)-477-7303 Fax: (305)-477-0013 

http://www.systeamusa.com

 

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.138 / Virus Database: 270.5.12/1595 - Release Date: 8/6/2008
8:23 AM

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Re: [asterisk-users] Strange beep during calls

[Alan Lord]

Felippe Silvestre wrote:
 Hi all,
  
 Our users are complaining about beeps that happen in the middle of some 
 calls. They are similar to the sound heard you are in a call and press 
 any button in your phone. Please find bellow some examples of these 
 beeps(the recordings are in Portuguese, but the beeps are easy to identify):
  
 http://www.katizak.locaweb.com.br/asterisk/beep.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3
 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3
  
 We are sure that our users are not pressing any button in the softphones 
 during the conversations.
 Do you guys are able to identify where these beeps are coming from? 
 Maybe an * functionality that we need to turn off... We are using 
 Asterisk 1.4.21.2.
  
 Thanks.
  

There was a short discussion on the OSLEC mailing list very recently 
about something that sounds (forgive the pun) similar. (Sorry I can't 
add anything else but I deleted them.)

I can't recall what the suggestions were but I think someone mentioned 
possible hardware faults on an analogue line card...

Al
-- 
The way out is open!
http://www.theopensourcerer.com

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Re: [asterisk-users] Strange beep during calls

[Tim Nelson]

I've also seen systems where the IRQ between the card and another heavily 
loaded device (disk controller) are shared causing clicks, beeps, and pops to 
be present in the audio stream.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- Alan Lord [EMAIL PROTECTED] wrote:

 Felippe Silvestre wrote:
  Hi all,
   
  Our users are complaining about beeps that happen in the middle of
 some 
  calls. They are similar to the sound heard you are in a call and
 press 
  any button in your phone. Please find bellow some examples of these
 
  beeps(the recordings are in Portuguese, but the beeps are easy to
 identify):
   
  http://www.katizak.locaweb.com.br/asterisk/beep.mp3
  http://www.katizak.locaweb.com.br/asterisk/beep2.mp3
  http://www.katizak.locaweb.com.br/asterisk/beep3.mp3
  http://www.katizak.locaweb.com.br/asterisk/beep4.mp3
   
  We are sure that our users are not pressing any button in the
 softphones 
  during the conversations.
  Do you guys are able to identify where these beeps are coming from?
 
  Maybe an * functionality that we need to turn off... We are using 
  Asterisk 1.4.21.2.
   
  Thanks.
   
 
 There was a short discussion on the OSLEC mailing list very recently 
 about something that sounds (forgive the pun) similar. (Sorry I can't
 
 add anything else but I deleted them.)
 
 I can't recall what the suggestions were but I think someone mentioned
 
 possible hardware faults on an analogue line card...
 
 Al
 -- 
 The way out is open!
 http://www.theopensourcerer.com
 
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Re: [asterisk-users] Max amount of concurrent calls on and iax trunk

[Chris Brentano]

I have two Asterisk 1.4 boxes connected via IAX over a VPN tunnel on a  
10Mbit link. We never did any stress testing as it's a temporary  
arrangement, but we've never had any call quality issues or run up  
against concurrent call limitations. I'm mostly routing internal  
extensions over the trunk, and in the case of two floating users I  
have their extensions at each office ring when their DID is called.  
One server is an older Pentium 4 1.7 GHz with 1GB Ram, and the other  
is a Dual Xeon 2.33 GHz with 4GB Ram. As for codec, I'm disallowing  
all except ulaw and gsm, with ulaw the priority codec for hardphones  
(Polycom) and gsm the priority for softphones (X-Lite, Zoiper).

I would expect the limitation you're going to run up against is not  
Asterisk, but the bandwidth between your two systems.


On 6 Aug, 2008, at 10:40 AM, Rosli Sukri wrote:

 hi,
 wanted to ask if anybody has experienced setting up two asterisk 1.2  
 boxes connected via iax trunk. have u guys ever stress tested the  
 trunks i.e how many concurrent calls can a trunk handle and whether  
 codec has any effect on it.
 ATT1.c


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[asterisk-users] Trying to understand Messages from chan_zap.c

[Daniel - Asterisk]

Hi friends,

Where can I get some information to understand messages like the following
ones?

*NOTICE[6455] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1*
*NOTICE[6455] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1*
* ERROR[6455] chan_zap.c: !! Got S-frame while link down*
*ERROR[6458] chan_zap.c: !! Got reject for frame 27, retransmitting frame 27
now, updating n_r!*
*ERROR[6458] chan_zap.c: !! Got a UA, but i'm in state 7*
*ERROR[6455] chan_zap.c: ACK received for '1' outside of window of '0' to
'0', restarting*
* ERROR[6455] chan_zap.c: !! Not good - head of queue has not been
transmitted yet*
*ERROR[6457] chan_zap.c: !! Got reject for frame 52, but we only have
others!*

Yesterday I have had several voice problems with my calls but my telephone
provider says he had no problems then but I don't think so. Today everything
was fine and I didn't do any change.

I'm using Asterisk 1.4.21.1 (very good with queue delivery), Zaptel 1.4.11
and Digium T412P card.

Thanks in advance!

Daniel
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[asterisk-users] Capture digits, set as variable..., use for caller id?

[Positively Optimistic]

We've searched but thus far have not successfully found a solution for this…

We're looking for a way to set a variable using get digits for a DISA
application.   Sometimes we're away from the office and get a voicemail that
I need to respond to quickly and would prefer for the caller to be presented
with the caller id of the office, or perhaps home….

I would like to set up DISA so that we can dial into the switch, enter a
password, provide the outgoing caller ID that we want to present, enter the
number I want to dial, and PRESTO..   make a call…   Any ideas?
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Re: [asterisk-users] asterisk realtime user deletion

[Nhadie]

i apologize, coz i had some experiences that my mail did not go thru. 
again i apologize.

Tilghman Lesher wrote:
 On Wednesday 06 August 2008 15:07:03 Nhadie wrote:
 Would just like to know if anyone has encountered this:
 
 You sent the exact same email this morning at 7:47 a.m.  If nobody has
 responded, it's because nobody has ever seen that before.  Duplicating
 the message tells everybody on the list that you have low regard for their
 time and bandwidth.
 

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[asterisk-users] Randulo: An open suggestion for the VOIP users Conference

[Karl Fife]

Randy:
Kudos to you for running the outstanding VOIP user's conference.   I
have an idea to toss into this public forum. I'm hoping that you and
others will consider it  give some feedback.  The idea would be to
begin each show with comments  corrections from the previous week's
show.  Sometimes when I listen to the previous week's archive, I find
that there is misinformation that could be corrected, or even a
'dangling' question that nobody on the conference could answer.  Someone
participating in the FOLLOWING week may be more inclined to comment,
correct, and even expand on such things if there were an official place
for it.  It may also make the first part of the show more dense with
specific useful information rather than being more free-form.  Doubtless
someone going through the archives would look forward to the beginning
of the NEXT archive which would start off with dense ( corrected) key
points of the previous call.

Example:   Last week there was talk about Polycom's HDVoice
technology, and the term was being used interchangeably with G.722.  In
fact there are important distinctions, but someone listening might
presume that the information was correct and leave short-changed.  There
are other examples even from last week, one involving someone's claim
that there's not a way to pick up a phone and directly interface with a
voice recognition directory application without needing to press some
digits first.  As it turns out, it's easy if you know the trick.  

Id' be happy to put my money where my mouth is and kick off this
Friday's show with these examples  any others I'm not remembering at
this moment if you think it would be well received.  Perhaps others will
do the same.
What do you think?  

Thanks!
-Karl Fife

If you want to discuss this off-list, you can email me at
[EMAIL PROTECTED]

p.s.
As it turns out, HDVoice CAN use G.722, but it can also be overlain onto
other codec's such as use G.722.1 and even G.711µ [sic].  That's right,
you can have an HDVvoice call over the PSTN using G.711, using a
special companding overlay on top G.711.  As I understand it, the two
HDVoice compliant endpoints (Polycom, Cisco  others that license the
technology) have an in-band (but inaudible) handshake, and then begin
applying the proprietary companding overlay which extends the dynamic
range of the audio.  It sounds great even though the underlying codec is
not a wideband codec.  Certainly the sound is not as good as HDVoice
over a modern adaptive-transform codec like G.722 (1987) or even better
over G.722.1 (1999), but it's definitely a big improvement over the
Toll-Quality (Read: AM-Radio-Through-A-Pillow) that we're all used to,
and it is not dependent upon having a pure-IP connection involving ENUM,
DUNDI, or other non e.164 namespaces such as SIP URI's, ITAD Subscriber
Numbers etc.  In my opinion HDVoice is it's a brilliant transition
technology.  

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