Steven J. Garcia
2005-Mar-18 18:35 UTC
[Asterisk-Users] T100P: Can't Make/Receive Zap Calls (Long Newbie Blah)
All,
Alright, I've looked around the internet, the voip-info.org wiki, and
browsed the contents of this mailing list. While I've found a couple of
scenarios that are close to this one, I haven't found one that uses my
particular card (T100P). Without further delay --
I have successfully configured internal SIP services between a Snom 200
and a Windows X-Lite client and have even connected to the Digium PBX
demo box through SIP (not that it makes much of a difference in this case).
So, what I'm trying to do is allow any internal SIP extension to dial
the outside world. I'd like to have eight dedicated Zap channels for
making outside calls, but I need to make Asterisk see them first! For
some reason, Asterisk refuses to see the PRI, but the T100P sees the
T1/PRI (I think).
We have the phone numbers we're to use assigned and I verified all the
signalling, framing, etcetera, with the provider.
I know lots of the stuff in extensions.conf is probably not needed, but
I wanted to include everything that I'm having the PBX box use. While
I've managed to get SOME of it working, I don't understand everything
about Asterisk.
Once again, thanks for help and sorry for the huge post. If I'm on
crack, please feel free to let me know.
Steven J. Garcia
IT Support
SLM Industries, L.L.C.
======================================================================Scenario/Problem
Data
======================================================================
Scenario: I'm responsible for configuring our office PBX which will
consist of the following: four individual voice lines, a conference
line, and a fax line. We currently have a T1 PRI through our service
provider (E8ZS/ESF; National/PRI_CPE). Our interfaces configuration is
set up as below
T100P -> Cisco IAD-2400 (PBX Port) -> Provider's Network ->
Provider's
Softswitch
// ++ I have compiled all of the required files for the following linux
// ++ distribution, kernel, gcc version (reported using cat
// ++ /proc/version).
Linux version 2.6.8-1-386 (joshk@trollwife) (gcc version 3.3.5 (Debian
1:3.3.5-2)) #1 Thu Nov 25 04:24:08 UTC 2004
I have compiled and installed the required sources (libpri, zaptel,
asterisk) in the listed order. I installed all other required libraries
as best as I could (and before the previously mentioned three). (Since
I'm not using RH, some of these packages had to be nabbed as best as
possible.) I have, to my knowledge, followed all the directions
required to make these functions work.
=====================================================================
Here is the error message that Asterisk throws when I try to connect to
an outside line.
pbx:/etc/asterisk# asterisk -vvvvvvvvvvvvvgr
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.2, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <markster@digium.com>
========================================================================Connected
to Asterisk 1.0.2 currently running on pbx (pid = 14307)
Verbosity was 5 and is now 13
-- Executing Dial("SIP/10.10.5.2-0811b1c0", "Zap/g1/<Real
7-digit
TN here>") in new stack
Mar 18 20:20:15 NOTICE[1148631984]: app_dial.c:743 dial_exec: Unable to
create channel of type 'Zap'
== Everyone is busy/congested at this time
-- Executing Congestion("SIP/10.10.5.2-0811b1c0", "")
in new stack
== Spawn extension (default, 9<Real 7-digit TN Here>, 2) exited
non-zero on 'SIP/10.10.5.2-0811b1c0'
pbx*CLI>
=====================================================================
// ++ ztcfg -vvvv returns the following output:
Zaptel Configuration
=====================
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: Individual Clear channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: D-channel (Default) (Slaves: 24)
24 channels configured.
// ++ And the light on the T100P is solid green.
============================================================
// ++ zttest returns the following output
pbx:/usr/src# /sbin/zttest -vvvv
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793%
--- Results after 4 passes ---
Best: 99.987793 -- Worst: 99.987793
=============================================================
// ++ cat /proc/zaptel/1 returns the following output
pbx:/usr/src# cat /proc/zaptel/1
Span 1: WCT1/0 "Digium Wildcard T100P T1/PRI Card 0" B8ZS/ESF
IRQ misses: 761
1 WCT1/0/1 Clear
2 WCT1/0/2 Clear
3 WCT1/0/3 Clear
4 WCT1/0/4 Clear
5 WCT1/0/5 Clear
6 WCT1/0/6 Clear
7 WCT1/0/7 Clear
8 WCT1/0/8 Clear
9 WCT1/0/9 Clear
10 WCT1/0/10 Clear
11 WCT1/0/11 Clear
12 WCT1/0/12 Clear
13 WCT1/0/13 Clear
14 WCT1/0/14 Clear
15 WCT1/0/15 Clear
16 WCT1/0/16 Clear
17 WCT1/0/17 Clear
18 WCT1/0/18 Clear
19 WCT1/0/19 Clear
20 WCT1/0/20 Clear
21 WCT1/0/21 Clear
22 WCT1/0/22 Clear
23 WCT1/0/23 Clear
24 WCT1/0/24 HDLCFCS
======================================================================
// ++ strace -xx cat /dev/zap/1 returns this and then lots of um-laued
// ++ (sp?) y-characters.
pbx:/usr/src# strace -xx cat /dev/zap/1
execve("/bin/cat", ["cat", "/dev/zap/1"], [/* 17
vars */]) = 0
uname({sys="Linux", node="pbx", ...}) = 0
brk(0) = 0x804d000
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0x40017000
access("/etc/ld.so.nohwcap", F_OK) = -1 ENOENT (No such file or
directory)
open("/etc/ld.so.preload", O_RDONLY) = -1 ENOENT (No such file or
directory)
open("/etc/ld.so.cache", O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=14363, ...}) = 0
old_mmap(NULL, 14363, PROT_READ, MAP_PRIVATE, 3, 0) = 0x40018000
close(3) = 0
access("/etc/ld.so.nohwcap", F_OK) = -1 ENOENT (No such file or
directory)
open("/lib/tls/libc.so.6", O_RDONLY) = 3
read(3, "\x7f\x45\x4c\x46\x01\x01\x01\x00\x00\x00\x00\x00\x00\x00"...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=1253924, ...}) = 0
old_mmap(NULL, 1260140, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x4001c000
old_mmap(0x40145000, 32768, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED,
3, 0x129000) = 0x40145000
old_mmap(0x4014d000, 10860, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x4014d000
close(3) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0x40150000
set_thread_area({entry_number:-1 -> 6, base_addr:0x401502a0,
limit:1048575, seg_32bit:1, contents:0, read_exec_only:0,
limit_in_pages:1, seg_not_present:0, useable:1}) = 0
munmap(0x40018000, 14363) = 0
open("/usr/lib/locale/locale-archive", O_RDONLY|O_LARGEFILE) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=290576, ...}) = 0
mmap2(NULL, 290576, PROT_READ, MAP_PRIVATE, 3, 0) = 0x40151000
close(3) = 0
brk(0) = 0x804d000
brk(0x806e000) = 0x806e000
brk(0) = 0x806e000
fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 0), ...}) = 0
open("/dev/zap/1", O_RDONLY|O_LARGEFILE) = 3
fstat64(3, {st_mode=S_IFCHR|0644, st_rdev=makedev(196, 1), ...}) = 0
read(3, "\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff"...,
4096) = 1024
write(1, "\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xff\xf
==================================================================
// ++ cat /proc/interrupts returns the following output
pbx:/usr/src# cat /proc/interrupts
CPU0
0: 628474284 XT-PIC timer
1: 203 XT-PIC i8042
2: 0 XT-PIC cascade
7: 11236543 XT-PIC parport0
8: 4 XT-PIC rtc
9: 0 XT-PIC acpi
10: 0 XT-PIC via82cxxx, uhci_hcd
11: 628967004 XT-PIC uhci_hcd, uhci_hcd, t1xxp, eth0
12: 479 XT-PIC i8042
14: 2086045 XT-PIC ide0
15: 12 XT-PIC ide1
NMI: 0
LOC: 628510985
ERR: 169
MIS: 0
pbx:/usr/src#
==============================================================
// ++ lsmod returns the following output
pbx:/usr/src# lsmod
Module Size Used by
ipv6 229764 10
snd_via82xx 26660 0
snd_ac97_codec 59268 1 snd_via82xx
snd_pcm 85384 1 snd_via82xx
snd_timer 23172 1 snd_pcm
snd_page_alloc 11144 2 snd_via82xx,snd_pcm
gameport 4736 1 snd_via82xx
snd_mpu401_uart 7296 1 snd_via82xx
snd_rawmidi 23204 1 snd_mpu401_uart
snd_seq_device 7944 1 snd_rawmidi
snd 50660 7
snd_via82xx,snd_ac97_codec,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device
wct1xxp 14752 0
zaptel 218500 3 wct1xxp
pci_hotplug 30640 0
via_agp 8832 1
agpgart 31784 1 via_agp
parport_pc 33348 0
parport 37320 1 parport_pc
mousedev 9996 0
floppy 54992 0
tsdev 7168 0
psmouse 17800 0
pcspkr 3816 0
evdev 9088 0
hisax 483280 0
isdn 128204 1 hisax
slhc 7040 1 isdn
uhci_hcd 29328 0
usbcore 104164 3 uhci_hcd
via82cxxx_audio 26248 0
uart401 11460 1 via82cxxx_audio
sound 75308 2 via82cxxx_audio,uart401
soundcore 9824 3 snd,via82cxxx_audio,sound
ac97_codec 16908 1 via82cxxx_audio
via_rhine 19720 0
mii 4864 1 via_rhine
crc32 4608 1 via_rhine
via_ircc 20368 0
irda 167360 1 via_ircc
crc_ccitt 2432 3 zaptel,hisax,irda
capability 4872 0
commoncap 7168 1 capability
ide_cd 38176 0
cdrom 35740 1 ide_cd
rtc 12088 0
ext3 109672 5
jbd 54552 1 ext3
ide_generic 1664 0
ide_disk 16768 7
via82cxxx 12956 1
ide_core 125028 4 ide_cd,ide_generic,ide_disk,via82cxxx
unix 25908 18
font 8576 0
vesafb 6688 0
cfbcopyarea 3840 1 vesafb
cfbimgblt 3200 1 vesafb
cfbfillrect 3712 1 vesafb
pbx:/usr/src#
==================================================================
// ++ Output of zap show channels from Asterisk Console
pbx*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default
pbx*CLI>
==================================================================
// ++ Output of pri show span 1 from Asterisk Console
pbx*CLI> pri show span 1
No PRI running on span 1
pbx*CLI>
==================================================================
// ++ Here is my hand-crafted zapata.conf, located in /etc/asterisk
pbx:/etc/asterisk# more zapata.conf
[Signalling Type]
#signalling=fxsks
signalling=pri_cpe
[ISDN PRI Switch Configuration]
switchtype=national
overlapdial=no
pridialpan=unknown
[Some Other Settings]
language=en
context=default
[Multilink PPP Options]
;minunused=2
;minidle=1
;idledial=6999
;idleext=6999@idle
[Analog Trunk Features]
usedistinctiveringdetection=no
;dring1=96,0,0
;dring2=325,95,0
;dring=367,0,0
;
;dring1context;dring2context;dring3context;
busydetect=yes
busycount=8
callprogress=no
pulse=no
[Analog Handset Features]
asdi=no
immediate=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
;
;group=1
;callgroup;
;group=1
;pickupgroup;
useincomingcalleridonzaptransfer=yes
[Caller ID Options]
callerid="SLM Industries <317 333 7900>"
calleridcallerid=asrecieved
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
[Audio Quality Options]
relaxdtmf=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
[Call Logging Options]
accountcode=slmindustries
amaflags=documentation
[Timing Parameters]
;prewink;preflash;wink;rxwink;rxflash;flash;start;debounce
[Other Features]
mailbox=4000
;
channel => 1-23
group = 1
[mrottler]
context=mrottler
[Call Logging Options]
accountcode=mrottler
amaflags=documentation
[Caller ID Options]
callerid="SLMI - M. Rottler <317 123-4567>"
calleridcallerid=asrecieved
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
[Other Features]
mailbox=4001
;
;group
[Analog Handset Features]
;
;group=1
;callgroup;
;group=1
;pickupgroupchannel => 2
[arottler]
context=arottler
[Call Logging Options]
accountcode=arottler
amaflags=documentation
[Caller ID Options]
callerid="SLMI - A. Rottler <317 123-4567>"
calleridcallerid=asrecieved
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
[Other Features]
mailbox=4002
;
;group
[Analog Handset Features]
;
;group=1
;callgroup;
;group=1
;pickupgroupchannel => 3
[rparker]
context=rparker
[Call Logging Options]
accountcode=rparker
amaflags=documentation
[Caller ID Options]
callerid="SLMI - R. Parker <317 123-4567>"
calleridcallerid=asrecieved
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
[Other Features]
mailbox=4003
;
;group
[Analog Handset Features]
;
;group=1
;callgroup;
;group=1
;pickupgroupchannel => 4
[sgarcia]
context=sgarcia
[Call Logging Options]
accountcode=sgarcia
amaflags=documentation
[Caller ID Options]
callerid="SLMI - S. Garcia <317 123-4567>"
calleridcallerid=asrecieved
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
[Other Features]
mailbox=4004
;
;group
[Analog Handset Features]
;
;group=1
;callgroup;
;group=1
;pickupgroupchannel => 5
=====================================================================
// ++ Here is my zaptel.conf, located in /etc
#[Wildcard T100P Configuration]
#[PRI Version]
#
# See the gathered documentation for more details on how to configure
# /etc/asterisk/zapata.conf with this configuration - Sar
#
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
#
#
======================================================================
// ++ Here is my obviously not handcrafted extensions.conf, located in
// ++ /etc/asterisk
[general]
static=yes
writeprotect=no
; The "Globals" category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variabl
e
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
;CONSOLE=Console/dsp ; Console interface for demo
CONSOLE=Zap/1
;CONSOLE=Phone/phone0
TRUNK=Zap/g1 ; Trunk interface
;
; please note that I have NO clue how to use the previous statement or
; how to organize groups, although it's probably insanely simple.
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => trunktollfree
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CH
ANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send t
o voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press
#, retur
n to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send
to voice
mail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press
#, retur
n to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as
no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press
*, send
the user into VoicemailMain
;
; My attempt to configure a sample extension for the Snom phone
; Hopefully this will work if I understand that instructions here well
enough.
;
[sgarcia_snomsip]
exten => 4004,1,Dial(SIP/sgarcia_snomsip,4004,20)
exten => 4004,2,Goto(4004-$(DIALSTATUS},1)
exten => 4004-NOANSWER,1,Voicemail(u4004) ; If
unavailable, send t
o voicemail w/ unavail announce
exten => 4004-NOANSWER,2,Goto(default,4004,1) ; If they
press #,
return to start
exten => 4004-BUSY,1,Voicemail(b4004) ; If busy, send
to voice
mail w/ busy announce
exten => 4004-BUSY,2,Goto(default,4004,1) ; If they
press #,
return to start
exten => _4004-.,1,Goto(4004-NOANSWER,1) ; Treat
anything e
lse as no answer
exten => a,1,VoicemailMain(4004) ; If they press *,
send the
user into VoicemailMain
include => local
include => longdistance
[sgarcia_vphone]
;
; Software Phone client configuration.
;
exten => 4404,1,Dial(SIP/sgarcia_vphone,4404,20)
exten => 4404,2,Goto(4404-$(DIALSTATUS},1)
exten => 4404-NOANSWER,1,Dial(SIP/sgarcia_snomsip,4004,20) ; If
unavailable,
dial my snom phone
exten => 4404-NOANSWER,2,Voicemail(b4004)
exten => 4404-NOANSWER,3,Goto(default,4404,1) ; If they
press #,
return to start
exten => 4404-BUSY,1,Voicemail(b4404) ; If busy, send
to voice
mail w/ busy announce
;exten => 4404-BUSY,2,Goto(default,4104,1) ; If they
press #
, return to start
exten => _4404-.,1,Goto(4404-NOANSWER,1) ; Treat
anything e
lse as no answer
exten => a,1,VoicemailMain(4004) ; If they press *,
send the
user into VoicemailMain
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,2,Goto(s,5) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold
while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,2,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,2,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try
again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call
the Aster
isk demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,4,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
include => sgarcia_snomsip
include => sgarcia_vphone
========================================================================
// ++ Please note that I am the Asterisk Ubernoob and am quite surprised
// ++ that I managed to get SIP working. Any recommendations/help on
// ++ this matter would be most appreciated.
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