First of all, before anybody starts telling me off, although I am a newbie, before posting this I did the following:
Learned linux basic commands
Performed succesful linux installs and configs
Installed and updated asterisk and the gui
After seeing how problematic gui is, did all editing by hand
Learned to use the CLI of asterisk
searched every major and minor source of voip info via google.
Now I am stuck. I have 5 extensions, 3 Cisco 7960’s and 2 softphones. They can talk to the asterisk. I am successfully registered at broadvoice. I was not sure how to confirm that, so I did sip show peers from the CLI and ran the GUI System tab. All seems to be connected.
My dialing plan is from broadvoice support site, and is simple:
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
NOW THE PROBLEM:
Trying to make an outgoing call, the CLI shows this and it is unsuccessful:
=========================================================================
Connected to Asterisk 1.6.1.0 currently running on localhost (pid = 20231)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
– Executing [17737103248@DLPN_DialPlan1:1] Dial(“SIP/707-08fc1540”, “SIP/17737103248@sip.broadvoice.com,30”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[May 24 05:23:50] WARNING[20794]: chan_sip.c:4630 sip_call: No audio format found to offer. Cancelling call to 17737103248
– Couldn’t call 17737103248@sip.broadvoice.com
== Everyone is busy/congested at this time (0:0/0/0)
– Executing [17737103248@DLPN_DialPlan1:2] Congestion(“SIP/707-08fc1540”, “”) in new stack
== Spawn extension (DLPN_DialPlan1, 17737103248, 2) exited non-zero on 'SIP/707-08fc1540’
localhost*CLI> – Executing [17737103248@DLPN_DialPlan1:1] Dial(“SIP/707-08fc1540”, “SIP/17737103248@sip.broadvoice.com,30”) in new stack
No such command ‘-- Executing [17737103248@DLPN_DialPlan1:1] Dial(“SIP/707-08fc1540”, “SIP/17737103248@sip.broadvoice.com,30”) in new stack’ (type ‘help – Executing’ for other possible
I HAVE TRIED. Please help if you can. is my dialplan messed up? What?
makes me think it is a problem with your audio codecs.
In your sip.conf you should have a line that says ‘disallow=all’ then a line under that that says ‘allow=ulaw’. ulaw is the most basic codec, it sounds good, but uses the same bandwidth as an analog telephone line(1/24th of a t-1). Your dialplan looks fine. If you keep having trouble post your sip.conf and we should be able to get it figured out.
You also might check what codecs broadvoice allows (g729, ulaw, gsm, etc).
In the general section was disallow=all and allow=ulawe,glaw,gsm.
I changed the allow to be allow=ulaw. That did not help. I stripped all the commands out of my sip.conf to make it easier to read, but I do not see the problem. Here is the file, I blanked the passwords:
[authentication]
[general]
;allow = ulaw,alaw,gsm
allow=ulaw
allowoverlap = no
allowexternaldomains = no
allowguest = no
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
authname = 212BLAHBLAH
autodomain = no
bindaddr = 0.0.0.0
bindport = 5060
callevents = no
canreinvite =
checkmwi = 10
compactheaders = no
context = default
defaultexpiry = 120
disallow = all
domain =
dtmf = inband
dtmfmode = rfc2833
dumphistory = no
externrefresh = 10
fromdomain =
fromuser = 212BLAHBLAH
g726nonstandard = no
host = sip.broadvoice.com
insecure = very
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize = ,
jbresyncthreshold = ,
language =
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
mwi_from =
nat =
notifyringing = no
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
register = 2122021963@sip.broadvoice.com:PASSWORD:2122021963@sip.broadvoice.com/707
registerattempts = 20
registertimeout = 30
relaxdtmf = no
rtpholdtimeout = 600
rtptimeout = 120
secret = PASSWORD
sendrpid = no
sipdebug = no
srvlookup = no
subscribecontext =
;tcpenable = no
tcpbindaddr = 0.0.0.0
tos_video = none
tos_audio = none
tos_sip = none
type = peer
udpbindaddr = 0.0.0.0
user = phone
useragent = Asterisk PBX
username = 212BLAHBLAH
usereqphone = no
videosupport = yes
t1min = 100
t38pt_udptl = no
trustrpid = no
You might try making your sip.conf as simple as possible. You have a ton of settings specified that probably do not need to be unless your provider asked you to set it this way.
Here is a very basic sip.conf, this setup does not show me registering to a sip provider, but you should only need that ‘register’ line and and the [sip.broadvoice.com] context. I am pretty sure the defaults for everything else would be fine.
also i dont think you need ‘dtmf’ just ‘dtmfmode’, you should be able to remove that line from [sip.broadvoice.com] and ‘user=phone’ looks unfamiliar this isnt a user or a phone, what is that setting doing? Perhaps remove that as well?
I did as you suggested, no luck. New sip.conf below. But I have been playing wioth this so long my head is spinning, and now I look and have to ask: shouldn’t a context references actually exist?
There is no context called from-broadvoice, yet in the sip.broadvoice.com it is referenced. Also the context “default.”
