Hi,
I haven't worked with Vonage myself but I usually get
this error back from my termination provider when the
number I have sent them is incorrect.
It might be worth checking you have used the correct
prefix (011 or 00) and area code etc.
Regards,
Aaron
=======Hi !
I have been working on making my asterisk server work
with Vonage
services. I have been able to recieve calls on my
asterisk machine but
i
couldnt call through that account to other people.
Means if i call a
zap
channel and then dial 1 314 652 ... then i get an
error like
Executing Dial("Zap/3-1", "SIP/<dialled
number>@sphone.vopr.vonage.net:5061")
in new stack
-- Called <dialled
number>@sphone.vopr.vonage.net:5061
-- Got SIP response 404 "Not Found" back from
216.115.25.198
-- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy
== Everyone is busy at this time
-- Executing Hangup("Zap/3-1", "") in new stack
== Spawn extension (local, 192512100488, 2) exited
non-zero on
'Zap/3-1'
-- Hungup 'Zap/3-1'
whether i dial any number ... i get the same
response... and always ...
Can anyone guess what might be the problem ?
in sip .conf my settings are :
register =>
<username>:<password>@sphone.vopr.vonage.net:5061
[sphone.vopr.vonage.net]
type = peer
fromuser = <username>
secret = <password>
host = <asterisk machine ip>:5070
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
nat = yes
canreinvite=no
In extensions.conf i have done :
exten =>
_1.,1,Dial,SIP/${EXTEN}@sphone.vopr.vonage.net:5061,tr
exten => _1.,2,Hangup
Please help me in this reagard.
Regards ,
Usman.
==========
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usman@user.iphonica.net
2004-Oct-19 04:15 UTC
[Asterisk-Users] Working Asterisk With Vonage
Hi !
I have been working on making my asterisk server work with Vonage
services. I have been able to recieve calls on my asterisk machine but i
couldnt call through that account to other people. Means if i call a zap
channel and then dial 1 314 652 ... then i get an error like
Executing Dial("Zap/3-1", "SIP/<dialled
number>@sphone.vopr.vonage.net:5061")
in new stack
-- Called <dialled number>@sphone.vopr.vonage.net:5061
-- Got SIP response 404 "Not Found" back from 216.115.25.198
-- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy
== Everyone is busy at this time
-- Executing Hangup("Zap/3-1", "") in new stack
== Spawn extension (local, 192512100488, 2) exited non-zero on
'Zap/3-1'
-- Hungup 'Zap/3-1'
whether i dial any number ... i get the same response... and always ...
Can anyone guess what might be the problem ?
in sip .conf my settings are :
register => <username>:<password>@sphone.vopr.vonage.net:5061
[sphone.vopr.vonage.net]
type = peer
fromuser = <username>
secret = <password>
host = <asterisk machine ip>:5070
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
nat = yes
canreinvite=no
In extensions.conf i have done :
exten => _1.,1,Dial,SIP/${EXTEN}@sphone.vopr.vonage.net:5061,tr
exten => _1.,2,Hangup
Please help me in this reagard.
Regards ,
Usman.
Looks like you're dialing on a zap channel, no?> -----Original Message----- > From: usman@user.iphonica.net [mailto:usman@user.iphonica.net] > Sent: Tuesday, October 19, 2004 6:15 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Working Asterisk With Vonage > > > Hi ! > I have been working on making my asterisk server work with Vonage > services. I have been able to recieve calls on my asterisk > machine but i > couldnt call through that account to other people. Means if i > call a zap > channel and then dial 1 314 652 ... then i get an error like > > Executing Dial("Zap/3-1", "SIP/<dialled > number>@sphone.vopr.vonage.net:5061") > in new stack > -- Called <dialled number>@sphone.vopr.vonage.net:5061 > -- Got SIP response 404 "Not Found" back from 216.115.25.198 > -- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy > == Everyone is busy at this time > -- Executing Hangup("Zap/3-1", "") in new stack > == Spawn extension (local, 192512100488, 2) exited non-zero > on 'Zap/3-1' > -- Hungup 'Zap/3-1' > > > whether i dial any number ... i get the same response... and > always ... > Can anyone guess what might be the problem ? > in sip .conf my settings are : > > register => <username>:<password>@sphone.vopr.vonage.net:5061 > > [sphone.vopr.vonage.net] > type = peer > fromuser = <username> > secret = <password> > host = <asterisk machine ip>:5070 > fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 nat = yes > canreinvite=no > > In extensions.conf i have done : > > exten => _1.,1,Dial,SIP/${EXTEN}@sphone.vopr.vonage.net:5061,tr > exten => _1.,2,Hangup > > Please help me in this reagard. > > Regards , > > Usman. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users
hallo, does anybody know, how to enable the the new iaxcomm client (with speex codec!!) to work with asterisk? i get a "Unable to find a path" error?? i have enabled speex in iax.conf, thanks for help, alex Nov 2 15:37:19 NOTICE[281616]: channel.c:1698 ast_set_write_format: Unable to find a path from GSM to SPEEX Nov 2 15:37:19 WARNING[281616]: file.c:779 ast_streamfile: Unable to open demo-echotest (format SPEEX): No such file or directory