* You are viewing Posts Tagged ‘trouble’

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, 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 office:

lsof -i:5060
COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME
asterisk 2045 asterisk 15u IPv4 23030 0t0 UDP *:sip

but:

telnet localhost 5060
Trying 127.0.0.1…
telnet: connect to address 127.0.0.1: Connection refused

iptables is set to allow 5060 udp and tcp. And I’ve flushed iptables,
but still no luck.

I can ssh into the office box from the home box. The office box is
directly connected, the home nat’ed.

Any help really appreciated.

sean

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 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 = 1402
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes

EXTENTION.CONF
[kantor-mtx]
exten => 1401,1,Dial(SIP/1401,60)
exten => 1402,1,Dial(SIP/1402,60)
exten => _NXXXNXXX,1,Dial(SIP/${EXTEN}@1401)
exten => _0813XXXXXXXX,1,Dial(SIP/${EXTEN}@1402)
exten => _0812XXXXXXXX,1,Dial(SIP/${EXTEN}@1402)

with this configuration files the result :
1.Sometime when i’m dialing to the PSTN line, not the PSTN ringing,
instead the extention is ringing
2.when I restart both asterisk and gxw4108, it’s succes when dialing
to PSTN, but when I try to dial the extention it’s seems dialing to
PSTN

(I also configuring to separate my incoming and outgoing call but
still the same error occurs)

my question is :
1.is it my gateway is broken or malfunction ?
2.When testing this configuration I’m using only 2 PSTN line instead
of 8 lines provide by the gateway, is this can cause problem in my
configuration ?
3.Or there is some other configuration for my asterisk and gxw4108,
instead the one I’m using would kindly to share in here

Thank you very much for your guidance and sorry if my English is bad

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:
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, if I test this account on a Linsys ATA it works just fine!)

Here is my sip.conf setting and the debug out put.

Thanks for help!

Jian

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 SIP/XXXX-00000404, strange things may happen.

Does anyone know what this warning means?

Thanks

Ish

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?

Thanks.

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 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 like
>
> core show uptime (not working)
> core restart (not present)
> core show version (not present)
> my Skype for Asterisk is also not loaded correctly.
>
> 190 modules are loaded, if I do a ‘module show’.
> I miss also some messages in the log like “[Jun 7 21:21:31]
> VERBOSE[3449] loader.c: func_version.so => (Get Asterisk
> Version/Build Info)”.
>
> Does anyone know something about this problem?
>
> best regards
> Christoph
>
>
> –
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