trouble with sip connection and registration

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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, 1.8.7.0
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

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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 192.168.14.80 and my grandstream IP is
192.168.101.184 here's my asterisk config files SIP.CONF
[1401]
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
secret…

Asterisk Users 3.9 years ago 0 Answers

SIP trunk trouble. Please help.

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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:
209.205.85.162
209.205.85.130 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.
(BTW,…

Asterisk Users 4.1 years ago 0 Answers

Trouble with *8 Pickup

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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 4.1 years ago 9 Answers

Macro to Dial a Channel Group using Round-robin

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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?
Thanks.

Asterisk Users 4.2 years ago 1 Answer

Update problem | CLI commands missing

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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 1.8.3.3.
>
> I have the following problem, if I do the update to the actual 1.8.4.2.
> There are several commands on the CLI which are not working or even
> not present…

Asterisk Users 4.3 years ago 0 Answers

call files

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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
agents...
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 4.4 years ago 3 Answers

No ringback even though progressinband=yes is set

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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 4.5 years ago 0 Answers

Asterisk -rx command not returning data - Version 1.4.33.1

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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 4.5 years ago 1 Answer

One Way Audio

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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 tim.compnetwork@gmail.com if you can help.

Asterisk Users 4.5 years ago 7 Answers