[Asterisk-Users] always 4 rings before * answers!?

[Ralf Illing]

Hi all,

 

I am trying to figure out how to configure asterisk to
pick-up immediatly.

I already had a look on the wiki
and google the lists ... I was experimenting around a
lot but still my phone rings exactly 4 times (always) before * answers it.
Anybody has an idea - or maybe sth. I just wrapped?

 

My Hardware: TDM400P PCI FXS, X100P PCI FXO

 

extensions.conf

exten =>
s,1,Answer

exten =>
s,2,setmusiconhold,default

exten =>
s,3,responsetimeout,10

exten =>
s,4,DigitTimeout,5

exten =>
s,5,Goto,language_menu|s|1

 

---zapata.conf

immediate=yes

 

Cheers

Ralf

 

 

Re: [Asterisk-Users] X100P Configs for Australia

[Ralf Illing]

We are using currently following settings: (melbourne)

 

Zapata.conf

--

echocancel=yes

echocancelwhenbridged=yes

echotraining=yes

rxgain=0.5

txgain=0.0

 

Works fine for us, but you can play of course a bit with the
gains …

 

 

-Ursprüngliche
Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Christopher Lee
Gesendet: Sunday, 18 January 2004
2:48 PM
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] X100P
Configs for Australia

 

Hi,

 

Just wondering if
anyone else in Australia is
using the X100P to connect to the PSTN, and what configs they have for it?

 

I’m finding
at present when I make a call I get a fair bit of echo of myself speaking, and
also the person on the other end can’t hear me very well (perhaps need to
up the digial Tx Gain? I don’t have it configured at present)

 

Asterisk is running
on Slackware Linux 9.1 and I built from the latest CVS just last night
(Saturday 17th Jan 04). The phone I’m using to call from is a
Cisco 7940 running the SIP 6.0 firmware. 

 

If I make calls
between the two Cisco 7940’s on my Asterisk system the voice quality is
fine.

 

The settings I have
for now are:-

 

Zapata.conf

=

[channels]

 

language=en

context=inbound-analog

signalling=fxs_ks

usecallerid=no

immediate=no

busydetect=no

callprogress=no

relaxdtmf=yes

echocancel=yes

echocancelwhenbridged=yes

callerid=asreceived

channel => 1

=

 

Zaptel.conf

=

fxsks=1

loadzone=au

defaultzone=au

=

 

Thanks,

Chris Lee

 

[Asterisk-Users] * dropps outbound calls over PSTN

[Ralf Illing]

Hi all,

I am experienced some strange dropping of outgoing calls, the time is
varying between 3 and 8 minutes before the call is dropped.
I have an Asterisk Developer's Kit (TDM) running on a gentoo box, Dell
OptiPlex with a P2 350. Most other stuff is working fine. I tried to
locate the problem, but the log files entries do not say much to me,
maybe somebody else has a bit more knowledge ...
Thanks in advance for any help

Cheers

Ralf

 

Feb 19 10:43:42 DEBUG[491538]: File chan_zap.c, Line 1461 (zt_call):
Dialing 'xx'
Feb 19 10:43:42 DEBUG[491538]: File chan_zap.c, Line 1507 (zt_call):
Deferring dialing...
Feb 19 10:43:43 DEBUG[491538]: File chan_zap.c, Line 3234
(zt_exception): Exception on 14, channel 1
Feb 19 10:43:43 DEBUG[491538]: File chan_zap.c, Line 2667
(zt_handle_event): Got event Hook Transition Complete(12) on channel 1
(index 0)
Feb 19 10:43:45 DEBUG[491538]: File chan_zap.c, Line 3234
(zt_exception): Exception on 14, channel 1
Feb 19 10:43:45 DEBUG[491538]: File chan_zap.c, Line 2667
(zt_handle_event): Got event Dial Complete(9) on channel 1 (index 0)
Feb 19 10:43:45 DEBUG[491538]: File chan_zap.c, Line 1064
(zt_enable_ec): Enabled echo cancellation on channel 1
Feb 19 10:43:45 DEBUG[491538]: File chan_zap.c, Line 2711
(zt_handle_event): Done dialing, but waiting for progress detection
before doing more...
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2014
(zt_setoption): Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/2-1
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2014
(zt_setoption): Set option TONE VERIFY, mode: MUTECONF/MAX(2) on Zap/1-1
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2290 (zt_bridge):
master: 2, slave: 1, nothingok: 0
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2305 (zt_bridge):
Stoping tones on 2/0 talking to 1/0
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2317 (zt_bridge):
Stoping tones on 1/0 talking to 2/0
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 2189 (zt_link):
Making 1 slave to master 2 at 0
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 906 (conf_add):
Added 14 to conference 9/2
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 906 (conf_add):
Added 15 to conference 9/1
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 1039 (update_conf):
Updated conferencing on 2, with 0 conference users
Feb 19 10:43:46 DEBUG[491538]: File chan_zap.c, Line 1039 (update_conf):
Updated conferencing on 1, with 0 conference users
Feb 19 10:46:28 DEBUG[491538]: File dsp.c, Line 1212 (ast_dsp_process):
Requesting Hangup because the busy tone was detected on channel Zap/1-1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 2138 (zt_unlink):
Unlinking slave 1 from 2
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 938 (conf_del):
Removed 14 from conference 9/2
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 938 (conf_del):
Removed 15 from conference 9/1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1039 (update_conf):
Updated conferencing on 2, with 0 conference users
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1048
(zt_enable_ec): Echo cancellation already on
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1048
(zt_enable_ec): Echo cancellation already on
Feb 19 10:46:28 DEBUG[491538]: File channel.c, Line 2226
(ast_channel_bridge): Returning from native bridge, channels: Zap/2-1,
Zap/1-1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1658 (zt_hangup):
Hangup: channel: 1 index = 0, normal = 14, callwait = -1, thirdcall = -1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1096
(zt_disable_ec): disabled echo cancellation on channel 1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 2025
(zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1039 (update_conf):
Updated conferencing on 1, with 0 conference users
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1658 (zt_hangup):
Hangup: channel: 2 index = 0, normal = 15, callwait = -1, thirdcall = -1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1096
(zt_disable_ec): disabled echo cancellation on channel 2
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 2025
(zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/2-1
Feb 19 10:46:28 DEBUG[491538]: File chan_zap.c, Line 1039 (update_conf):
Updated conferencing on 2, with 0 conference users
Feb 19 10:46:35 DEBUG[114696]: File chan_zap.c, Line 1096
(zt_disable_ec): disabled echo cancellation on channel 2
Feb 19 10:46:47 DEBUG[114696]: File chan_zap.c, Line 1064
(zt_enable_ec): Enabled echo cancellation on channel 2
Feb 19 10:46:47 DEBUG[507922]: File chan_zap.c, Line 1941 (zt_answer):
Took Zap/2-1 off hook
Feb 19 10:46:48 DEBUG[507922]: File channel.c, Line 953
(ast_settimeout): Scheduling timer at 160 sample intervals
Feb 19 10:46:50 DEBUG[507922]: File chan_zap.c, Line 3234
(zt_exception): Exception on 15, channel 2
Feb 19 10:46:50 DEBUG[507922]: File chan_zap.c, Line 2667
(zt_handle_e

Re: [Asterisk-Users] * dropps outbound calls over PSTN

[Ralf Illing]

Had busycount disabled, thus I put it to the standard value of 6, and
will have some test calls running over the day.
Somebody out there who has experiences with telstra vic/au - what the
best values here are?
Cheers
Ralf

> It appears that this was disconnecting because * thinks it detected a
> busy tone. 
> 
> Here is the important message from your log:
> 
>> Feb 19 10:46:28 DEBUG[491538]: File dsp.c, Line 1212
>> (ast_dsp_process): Requesting Hangup because the busy tone was
>> detected on channel Zap/1-1
> 
> I'm not very familiar with the busydetect options, but I would
> suggest disabling it or at least setting it higher 
> 
> James
> 

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[Asterisk-Users] Unable to specify channel 2: Device or resource busy

[Ralf Illing]

Hi folks,
 
I installed my Asterisk Developer's Kit (TDM) "successfully" on a old
Dell machine and ending up with some problems. 
 
Running my start script .
 
rmmod audio
rmmod wcfxo
rmmod zaptel
modprobe wcfxo
sleep 3
ztcfg -vvv
sleep 1
asterisk -gc
 
. ends in following problem:
 
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
2 channels configured.
 
ZT_CHANCONFIG failed on channel 2: No such device or address (6)
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[Asterisk-Users] Play sound to callee

[Ralf Illing]

Hi all,

 

I am setting up * for the first time, every thing is working
fine, but I would like to implement an additional feature:

Thus we have multilingual caller menu – I would like
to play a little sound file to the callees to let them know in which language they
should answer the incoming call, before I pass the caller through (caller
should hear the normal ringing meanwhile). Is that possible?

 

That’s the way I am putting the call through at the
moment

exten => s,1,Dial,Zap/2|20

 

Thx, for any help.

Cheers

Ralf

[Asterisk-Users] Sendmail not on localhost

[Ralf Illing]

Hi …

 

I already set-up sendmail on
another network server thus it would be nice to use that one or is sendmail on * server required!?

I had a look in the archive but couldn’t find any
information where to set the mail server from localhost
to my network server …

 

Cheers

Ralf