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 3.9 years ago 2 Answers

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 4 years ago 0 Answers

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 4.2 years ago 0 Answers

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 4.2 years ago 9 Answers

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 4.3 years ago 1 Answer

Update problem | CLI commands missing


Hi List, is there somebody how is able to help me here? Or at least to get more
details why this occurs? best regards
Christoph Am 08.06.2011 18:00, schrieb Christoph Timm:
> Hi List,
> I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW.
> Currently I'm running Asterisk
> I have the following problem, if I do the update to the actual
> There are several commands on the CLI which are not working or even
> not present…

Asterisk Users 4.4 years ago 0 Answers

call files


Hi. Im having trouble setting variables in channel dialplan and re-using them in
Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context: agents
Extension: 888210332450
Set: __PARTNER=ZonNew-Outbound
Set: NUMBER=210332450
- In "Local/210332450@ZonNew-Outbound" I Set(bla='blabla'); It seems I cannot re-use this var in extension _888XXXXXXXXX in context
Basically the Channel dialplan has a Queue() and in _888XXXXXXXXX I would
like to know the peer (or interface) that answered it... What can I…

Asterisk Users 4.5 years ago 3 Answers

No ringback even though progressinband=yes is set


Steve. Thanks for the insight. I won't pretend to know what "early-audio" is, but I guess I'm about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 & 1.6 in this scenario? Thanks a million!! :-) -
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300

Asterisk Users 4.6 years ago 0 Answers

Asterisk -rx command not returning data - Version


Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload'
siptest:~# and a "verbose 10" setting shows [Mar 14 19:07:41] ERROR[3092]: utils.c:968 ast_carefulwrite: write()
returned error: Broken pipe

Asterisk Users 4.6 years ago 1 Answer

One Way Audio


I am having trouble with no return audio on inbound calls. I have been
working on this for 18 hours and even built a fresh server and moved
everything over and I am getting the same results. I need someone that can
help get this resolved tonight. I can provide access to all machines
involved. Please email me at if you can help.

Asterisk Users 4.6 years ago 7 Answers