Janusz Starzyk
2004-Feb-24 15:26 UTC
[Asterisk-Users] painful first steps with * and SIP phones
I hope, you do not mind few questions from the newbie:
As a first after installing the Asterisk I try to make it working with two
PCs and Xten-Lite installed.
Both PCs have fixed IP addresses but they log in to Asterisk only if
configured in sip.conf with host=dynamic. Why?
I have them defined in the default context in extensions.conf as:
exten => 621,1,Dial(SIP/621,20)
exten => 621,2,Voicemail(u621)
exten => 621,3,Hangup
exten => 621,102,Voicemail(b621)
exten => 621,103,Hangup
exten => 622,1,Dial(SIP/622,20)
exten => 622,2,Voicemail(u622)
exten => 622,3,Hangup
exten => 622,102,Voicemail(b622)
exten => 622,103,Hangup
in the sip.conf I put for each sip phone the following (<x> equals 1 and 2
respectively)
[62<x>]
type=friend
secret=<password>
host=dynamic
username=62<x>
dtmfmode=inband
mailbox=62<x>
callerid="<name>"
context=default
The Xten-Lite has dtmfmode set to inband too.
When I try to call from one phone the other one, I get voicemail saying the
other party is unavailable.
Moreover, I can not navigate in the voicemail as the key press seems to be
not received or unrecognized by the voicemail.
ON the Asterisk console I get the following messages:
Executing Dial("SIP/621-2cea","SIP/622|20") in new stack
Specify address with base=0xNNNNN
Called 622
Feb 24, WARNING[-1126650960]: chan_sip.c:495 retrans_pkt: Maximum retries
exceeded on call ........ for seqno 102 (Request)
No one is available to answer at this time
Executing Voicemail......
Playing 'vm-theperson'
Feb 24 WARNING[-1231549520]: dsp.c:1424 ast_dsp_process: Unable to process
inband DTMF on 2 frames -- repeated many times
Can somebody please help and suggest any corrections?
Thanks,
Janusz
Jan Larsen
2004-Feb-24 16:44 UTC
[Asterisk-Users] painful first steps with * and SIP phones
Hi
I have the same setup as You. Here are my sip.conf , extensions.conf and
modules.conf
that is currently on my system.
Just copy and paste, change numbers and reload..........
This is my sip.conf which I have on my system and it is working. It is a
modified eksample file
; 2003-04-24 05:06 GMT jtodd@loligo.com
;
; This is the sip.conf file for John Todd's Asterisk
; server. Asterisk can be found on http://www.asterisk.org/
;
; More recent versions of this file can be found on:
; http://www.loligo.com/asterisk/
;
;
; SIP Configuration for Asterisk
;
[general]
;disallow=gsm
;allow=ulaw
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = local ; Default for incoming calls
callerid=No CallID
; Other general settings for the SIP channels that are used
;
; tos=['lowdelay', 'throughput', 'reliability',
'mincost', or 'none']
; Sets Type Of Service flags (?)
;
; maxexpirey=3600
; Max duration (seconds) of incoming registration allowed
;
; defaultexpirey=120
; Default length of incoming/outoing registrations
;
;
; Under each peer/user/friend, you can specify some additional notes:
; tos=['lowdelay', 'throughput', 'reliability',
'mincost', or 'none']
;
; My SIP phones in the house/office are listed below
;
; By my own convention (and to save my sanity) I have given
; all the ATA-186 devices in my network "extensions" that
; are numeric. I picked "2200" to start for historical
; laziness. Again, for my sanity, I also made the usernames
; on the phone identical to the extension number of the
; channel.
;
; I distribute IP addresses with a DHCP server to these phones,
; on the network 204.91.156.0/24 or behind a NAT (for some)
;
; All the Cisco ATA-186 units I have are on "real" IP addresses,
; and I haven't experimented with NAT yet. NAT has been a real
; PITA with the ATA's, and I don't relish having to make it work.
;
; Anytime a call is dialed from "2203" or "2204", it will be
; parsed by the context [intern] in the extensions.conf file
;
; If you want the message waiting light to light up, or stutter
; dialtone, you can set mailbox=[mbox#] - note you can also
; set a comma-delimited list of mailboxes if you want multiples.
;
; You can use "permit=x.x.x.x/yy" lists for hosts that are dynamic
;
; You don't need to register SIP peers that are static (host=)
;
; Setting "nat=1" is required to get the "Via:" headers to
be configured
; correctly so that the Ciscos (and others?) will re-register when
; they are behind a NAT. If nat=1 is set, it will not harm non-NAT
; connections on the Ciscos, but there are reports of other phones
; (pingtel) not working if you have nat=1 set and the phone is not
; behind a NAT.
;
[2201] ; X-Lite client 2201
type=friend
secret=<password>
auth=md5
nat=1 ; we assume clients are behind
NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=rfc2833 ; To be used with dub-fix.diff to avoid
dubble dtmf tones
callerid="any name" <2201>
disallow=all
allow=ulaw ; since we chose 'inband' for
dtmf
we must use g.711
allow=alaw ; add whatever other codecs we
fancy
context=local ; use a context that exists ;-)
mailbox=2201
[2202] ; X-lite client 2202
type=friend
secret=<password>
auth=md5
nat=1 ; we assume clients are behind
NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=rfc2833 ; To be used with dub-fix.diff to avoid
dubble dtmf tones
callerid="any name" <2202>
disallow=all
allow=ulaw ; since we chose 'inband' for
dtmf
we must use g.711
allow=alaw ; add whatever other codecs we fancy
context=local ; use a context that exists ;-)
mailbox=2202
This is my extensions.conf which I have on my system and it is working. It
is a modified eksample file
; 2003-04-24 05:06 GMT jtodd@loligo.com
;
; This is the extensions.conf file for John Todd's Asterisk
; server. Asterisk can be found on http://www.asterisk.org/
;
; More recent versions of this file can be found on:
; http://www.loligo.com/asterisk/
;
;
; The "General" category is for certain variables. All other
categories
; are interpreted as extension contexts
;
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
; if stati=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
[general]
static=yes
writeprotect=yes
; The [globals] context is where you can set variables that
; can be referenced elsewhere in the dialplan with ${VARIABLE}
;
; I decided that for ease of reference, I should create a variable
; called "PHONE1" that I could set to the phone where I normally
; am found. I then set "ME" to be my extention, raw, for use
; with voicemail forwarding.
;
; Variable "PHONE2" is the other ATA-186 in the house.
; I create variables here so that if I decide to update my
; extensions list, or my dial-out interface list, it's just
; a simple variable change here at the top of the file.
;
[globals]
PHONE1=SIP/2201
PHONE1VM=2201
PHONE2=SIP/2202
PHONE2VM=2202
TRUNK=CAPI
; The [intern] context is where I pass all calls
; that are dialed by the SIP phones in my house/office.
; I name these phones in the sip.conf file when I define the
; SIP peers.
;
; This context is the "main" context that gets used for outbound
; calls out of the house. It, in turn, calls other contexts
; which contain bits and pieces of other places that it's possible
; to dial from my house phone. I've split them up into many sub-segments
; so that they are re-usable if I should decide to allow other
; persons or organizations access some of those dialplan parts
; in some future configuration. My older extensions.conf had many
; fewer contexts, but I have opted to build this version with more
; flexible use of "include" commands to make my life easier for
; future configuration chores.
;
;[intern]
;exten => _.,1,NoOp
;exten => _.,2,Goto(intern-post,${EXTEN},1)
; After we set up some initial housekeeping things in the [intern] context,
; the [intern-post] context is jumped to, which is where the true dialplans
; are kept.
;
;[intern-post]
; Send calls to operator (private PSTN gateway, analog line in my house)
;include => operator
; Send 911 calls to emergency outbound (private PSTN gateway, analog line in
my house)
;include => e911
; If someone dials a "9" in front of their number, send out via analog
(private PSTN gateway)
;include => forced-analog
; If someone dials one of the 22xx numbers that are local (SIP phones in my
house)
;include => local
;[operator]
; Handle calls to the Operator ("9")
;
; All Operator calls go to the analog line, since
; that is the only place that there is an operator
; on duty. At some point, this extension will lead
; a particular person in the office.
;
; Note that one can reach the operator by dialing "90"
; according to the pattern sets below, but I include
; just a single "9" as this system will be used by
; persons who are not used to dialing prefixes for
; their phone calls.
;
; I have to build an international dialing plan
; shortly to use a different iconnect account, which
; has international rate plans. Currently, the system
; simply does not allow interational dialing (011...)
; unless it's sent out to the local analog via the "9"
; prefix below.
;
;exten => 9,1,Dial(${DIALOUTANALOG}/${EXTEN},70)
;exten => 9,2,Macro(fastbusy)
;exten => 9,102,Playback(ss-noservice)
;exten => 9,103,Macro(fastbusy)
;[e911]
; EMERGENCY OUTBOUND
;
; United States dialing supports "911" emergency calls
; which are routed to the "closest" emergency contact center.
;
; Of course, you need to make sure that the center to which
; you are connecting is actually able to do something useful
; for the dialer; it doesn't make sense to have your NY branch
; offices dialing a 911 center in San Francisco.
;
; Note: The ATA-186 is auto-configured to send "911"
; directly to the SIP server with no delay. Be very, very
; careful when experimenting with this - the system does not
; follow standard rules with 911 calls.
;
; If you're !(USA), feel free to comment this context out entirely.
;
;exten => 911,1,Dial(${DIALOUTANALOG}/${EXTEN})
;exten => 911,2,Macro(fastbusy)
;[forced-analog]
; This is for failure-case outbound dialing. If for some reason
; the outbound SIP connections aren't working, but aren't giving errors
; (i.e.: my path to larger Internet is dead) then I can short-circuit
; my way to the local analog dial port by dialing "91npaxxxyyyy" and
; the call will be shunted out the analog port. This is for when
; the inevitable cry of "IT DOESN'T WORK!!" is screamed from the
; living room by She Who Makes Calls.
;
;exten => _9.,1,Dial(${DIALOUTANALOG}/${EXTEN:1},70,Tt)
;exten => _9.,2,Macro(fastbusy)
;exten => _9.,102,Macro(fastbusy)
; The [inbound-analog] context is where calls coming in from the
; zaptel X100P analog circuit are passed. Right now, this
; is a very simple dial plan that rings my two SIP phones.
;
; First, send the call through Zapateller. If there's no callerid,
; the system will answer the line, play a quick two-tone "re-order"
; signal which will cause many telemarketer systems to register the
; number as "out of order" and remove it from their lists. If the
; caller is a human, they'll jump to the next step...
;
; Next, if desired, start recording the call with the record-on
; macro. See my notes on legality. Make sure to clean up the
; calls with the record-cleanup macro in the "h" extension.
; Since I don't have the "called" number at this point (I only
; have "s") I will fill in what normally is ${EXTEN} in the
; macro call with the static ${MYHOMEPHONE} variable.
; Recording commented out and replaced by NoOp.
;
; The PrivacyManager will only work if the caller has no caller ID
; information in the inbound call. The PrivacyManager application
; will then give the caller 3 attempts to enter in a 10 digit phone
; number. That number then becomes the CallerID, and the call
; process is continued.
;
; Ring both SIP phones in the house that are "general" lines.
;
; If the lines ring for >20 seconds, play an amusing message and
; then dial the "u"navaialable voicemail box.
; If the lines are busy, dial the "b"usy voicemail box.
;
; See the file zapata.conf for how to configure the Zaptel X100P
; to forward calls to this context when called
;
; Handels incomming call on ISDN Bri
[incomming-ISDN2]
exten => s,1,NoOp
exten => s,2,Dial(${PHONE1},35,Ttm)
exten => s,3,Voicemail(u${PHONE1VM})
exten => s,104,Hangup
; The [local] context is where I put all of the locally dialable
; numbers. These entries could be made in each context where I
; reference them, but that would be redundant.
;
; Mostly, they are used from context [intern] right now, but I'm sure
; I'll include them in other contexts later.
;
; Note: due to my wacky/kludgey prefix dialing crap, all the
"internal"
; extensions that are local to this system must begin with a "2"
; This means that voicemail, echo, all users, etc. must have an extension
; that starts with the digit "2" or one of the other dialplans will
; snag the call and do whatever with it. This is sub-optimal for a
; system with hundreds of extensions, but for my purposes it works
; just fine.
; Furthermore, all of "my" extensions begin with 220, since I
don't
; expect to have more than 10 phones. My friends can then have
; extension plans with 221X or 222X or whatever, and we won't
; overlap.
;
;
; First extension (office) in the house
;
; Note that I try to keep extension names and numbers
; identical. You don't need to be bound to this
; method.
;
; Note that many internal extensions do not time out to voicemail.
;
[local]
;
; Test ISDN2 access
exten => _0XXX.,1,StripMSD,1
exten => _XXX.,2,Dial,CAPI/*:bBYEXTENSION
; 2201 is my test Phone
;
exten => 2201,1,Dial(${PHONE1},60,t) ; Ring for 60 seconds
exten => 2201,2,Voicemail(u${EXTEN}) ; Unavail voicemail if extension
doesn't answer
exten => 2201,3,Hangup
exten => 2201,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is busy
exten => 2201,103,Hangup
; 2202 is my test Phone
;
exten => 2202,1,Dial(${PHONE2},60,t) ; Ring for 60 seconds
exten => 2202,2,Voicemail(u${EXTEN}) ; Unavail voicemail if extension
doesn't answer
exten => 2202,3,Hangup
exten => 2202,102,Voicemail(b${EXTEN}) ; Busy Voicemail if extension is busy
exten => 2202,103,Hangup
; Dial 3000 from any phone to go to the voicemail system
;
exten => 3000,1,Wait,2
exten => 3000,2,VoicemailMain(${EXTEN})
exten => 3000,3,Hangup
This is my modules.conf which I have on my system and it is working. It is a
modified eksample file
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => res_musiconhold.so
;
; MUST to be loaded before chan_capi.so
load => res_parking.so
;
; Load capi driver
load => chan_modem_i4l.so
load => chan_capi.so
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
;noload => chan_alsa.so
noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
chan_modem.so=yes
res_parking.so=yes
chan_capi.so=yes
Regards
Jan Larsen
----- Original Message -----
From: "Janusz Starzyk" <jstarzyk@adres.pl>
To: <asterisk-users@lists.digium.com>
Sent: Tuesday, February 24, 2004 11:26 PM
Subject: [Asterisk-Users] painful first steps with * and SIP phones
> I hope, you do not mind few questions from the newbie:
>
> As a first after installing the Asterisk I try to make it working with two
> PCs and Xten-Lite installed.
> Both PCs have fixed IP addresses but they log in to Asterisk only if
> configured in sip.conf with host=dynamic. Why?
>
> I have them defined in the default context in extensions.conf as:
>
> exten => 621,1,Dial(SIP/621,20)
> exten => 621,2,Voicemail(u621)
> exten => 621,3,Hangup
> exten => 621,102,Voicemail(b621)
> exten => 621,103,Hangup
>
> exten => 622,1,Dial(SIP/622,20)
> exten => 622,2,Voicemail(u622)
> exten => 622,3,Hangup
> exten => 622,102,Voicemail(b622)
> exten => 622,103,Hangup
>
> in the sip.conf I put for each sip phone the following (<x> equals 1
and 2
> respectively)
>
> [62<x>]
> type=friend
> secret=<password>
> host=dynamic
> username=62<x>
> dtmfmode=inband
> mailbox=62<x>
> callerid="<name>"
> context=default
>
> The Xten-Lite has dtmfmode set to inband too.
>
> When I try to call from one phone the other one, I get voicemail saying
the> other party is unavailable.
> Moreover, I can not navigate in the voicemail as the key press seems to be
> not received or unrecognized by the voicemail.
>
> ON the Asterisk console I get the following messages:
>
> Executing Dial("SIP/621-2cea","SIP/622|20") in new
stack
> Specify address with base=0xNNNNN
> Called 622
> Feb 24, WARNING[-1126650960]: chan_sip.c:495 retrans_pkt: Maximum retries
> exceeded on call ........ for seqno 102 (Request)
> No one is available to answer at this time
> Executing Voicemail......
> Playing 'vm-theperson'
> Feb 24 WARNING[-1231549520]: dsp.c:1424 ast_dsp_process: Unable to
process> inband DTMF on 2 frames -- repeated many times
>
> Can somebody please help and suggest any corrections?
>
> Thanks,
> Janusz
>
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