trouble with sip connection and registration


I have a home asterisk box which connects to the office asterisk, so I
can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box,
on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout: --
Registration for 'sip@' timed out, trying again (Attempt #86) I first thought it was some fall out of the new upgrade to 10.0-rc1, but
now I'm not sure. Even with verbose at 6 I don't see anything on the office console about
the attempted registration. And on the…

Asterisk Users 3.7 years ago 2 Answer

Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment


Dear all,
I'm still having trouble using asterisk with the grandstream gxw4108,
in the gxw4108 I'm using 1 stage dialing in the profile1 I already
type my asterisk server address and my grandstream IP is here's my asterisk config files SIP.CONF
type = friend
username = 1401
secret = 1401
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes [1402]
type = friend
username = 1402

Asterisk Users 3.7 years ago 0 Answer

SIP trunk trouble. Please help.


Hello, I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5.
The problem is can't make any outbound/inbound. It always get "Number is
not valid 701". I tried to figure out the reason the call got dropped and couldn't find
out the solution. I noticed that in the SIP debug there are two IP (from
the provider) involved: It seems my Asterisk sent INVITE to the first IP, but the provider want
use 2nd one. How can I make it works? I never seen this thing before.

Asterisk Users 3.9 years ago 0 Answer

Trouble with *8 Pickup


We have a client that has sporadic problems with the *8 pickup facility.
The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they
get a forbidden message on the phone and I can see the following in the
logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
[Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL
[Aug 8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup of SIP/XXXX-00000404 failed.
[Aug 8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel…

Asterisk Users 4 years ago 9 Answer

Macro to Dial a Channel Group using Round-robin


Good morning, I am writing a Asterisk dialplan from scratch (for learning and
testing purposes), but i'm having trouble with a algorithm to dial a SIP
group using round-robin. I want that asterisk dial the member of the
group in a circular way, until the call be answered. For example, i have
the group TEST="SIP/1&SIP/2&SIP/3&SIP/4", asterisk would dial SIP/1, if
it doesn't answer in a period of time then asterisk would dial SIP/2 and
so on. Can somebody help me?

Asterisk Users 4 years ago 1 Answer