SIP interconnection

Asterisk Asterisk Support
1

Hello (again today !)

One issue concerning using a SIP provider for calling. I get the following in the CLI:

– Executing Dial(“SIP/opmob1-081f8de0”, “SIP/0442612880@freephonie”) in new stack
– Called 0442612880@freephonie
Dec 19 19:32:13 WARNING[2995]: chan_sip.c:9947 handle_response_invite: Forbidden - wrong password on authentication for INVITE to ‘“Operator1” sip:10001@192.168.1.100;tag=as022c3d31’
– SIP/freephonie-082203d0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
I don’t understand “wrong password on authentication for INVITE” because it concerns one of my hardphone which works fine. And finally why “circuit-busy” ?

any idea ?

Thx
Lilian

2

I have the same problems.
I try to make a trunk between to asterisk boxes, one of them can receive and make a call, but the other can’t. When a client in the problem’s asterisk make a call, the output in the CLI is :
.
.
== Everyone is busy/congested at this time
.
.

3

Hi
Plz post ur sip.conf peer & genreal context. Also check your out going call context in extension.conf sud be mention as context in type=friendmeans supporr ur user extension is 330 & he wants to dail out call then in sip.conf for that user
[330]
type=friend
context= out call context
this may be also issue
so check such context inclusion in sip.conf & extension.conf

Regards
Amit

4

I’m using a database to store sip general:

1|0|sip.conf|general|bindport|5060
2|0|sip.conf|general|bindaddr|0.0.0.0
3|0|sip.conf|general|srvlookup|no
4|0|sip.conf|general|language|fr
5|0|sip.conf|general|maxexpiry|3600
6|0|sip.conf|general|defaultexpiry|1800
7|0|sip.conf|general|useragent|My PBX
8|0|sip.conf|general|nat|no
9|0|sip.conf|general|qualify|yes
10|0|sip.conf|general|rtcachefriends|yes
11|0|sip.conf|general|allowguest|yes
12|0|sip.conf|general|tos|lowdelay
13|0|sip.conf|general|relaxdtmf|yes
14|0|sip.conf|general|context|from-sip
15|0|sip.conf|general|checkmwi|10
16|0|sip.conf|general|vmexten|*98
17|0|sip.conf|general|videosupport|no
18|1|sip.conf|general|realm|
19|1|sip.conf|general|dtmfmode|
20|1|sip.conf|general|externip|
21|1|sip.conf|general|localnet|
22|1|sip.conf|general|musicclass|
23|1|sip.conf|general|disallow|
24|0|sip.conf|general|allow|gsm,alaw,ulaw,g729
25|0|sip.conf|general|register|0123456789:my_password@freephonie.net:5060

and in my extension.conf I have:
[to-extern]
exten = _9.,1,Dial(SIP/${EXTEN:1}@freephonie)

and in my usersip table:
12|freephonie|0||||my_callerid|no|free-outgoing||rfc2833||||212.27.52.5|||||yes|||||5060|no||||my_password|friend|0950808439|all|gsm,g729,ulaw,alaw||0||||||trunk

structure of usersip table:
0|id|integer|0||1
1|name|varchar(80)|99||0
2|commented|tinyint(1)|99|0|0
3|accountcode|varchar(20)|0||0
4|amaflags|varchar(13)|0||0
5|callgroup|varchar(10)|0||0
6|callerid|varchar(80)|0||0
7|canreinvite|char(3)|0||0
8|context|varchar(80)|0||0
9|defaultip|varchar(15)|0||0
10|dtmfmode|varchar(7)|0||0
11|fromuser|varchar(80)|0||0
12|fromdomain|varchar(80)|0||0
13|fullcontact|varchar(80)|0||0
14|host|varchar(31)|99||0
15|insecure|varchar(11)|0||0
16|language|char(2)|0||0
17|mailbox|varchar(128)|0||0
18|md5secret|varchar(80)|0||0
19|nat|varchar(5)|99|no|0
20|deny|varchar(95)|0||0
21|permit|varchar(95)|0||0
22|mask|varchar(95)|0||0
23|pickupgroup|varchar(10)|0||0
24|port|varchar(5)|99||0
25|qualify|char(3)|0||0
26|restrictcid|char(1)|0||0
27|rtptimeout|char(3)|0||0
28|rtpholdtimeout|char(3)|0||0
29|secret|varchar(80)|0||0
30|type|varchar(6)|99|friend|0
31|username|varchar(80)|99||0
32|disallow|varchar(100)|0||0
33|allow|varchar(100)|0||0
34|musiconhold|varchar(100)|0||0
35|regseconds|integer|99|0|0
36|ipaddr|varchar(15)|99||0
37|regexten|varchar(80)|99||0
38|cancallforward|char(3)|0||0
39|setvar|varchar(100)|99||0
40|call-limit|tinyint|0|0|0
41|category|varchar(50)|99||0

thanks for your help
Lilian

5

Hi

 the sip table for genreal is ok

But the table format ur use for extension or users is wrong never use Varchar(10) this indicate placed name or character whos length is 10 so
plz check sip table examples on voip-info.org where u get idea how to develpoe sip table
And no need to seprate table for sip users
Also i am adivce u config only sip users in db table all genreal & peer paramter in actual sip.conf
As well context of type=peer & Type=friend means users are ur out call context which =to-extern
Also mention this in context colum for each user& Peer

all over ur sip user table is wrong & hence sip.conf config is wrong so correct it

Regards
Amit

6

Thanks for your support,

will try to cleanup my config

Thanks
Lilian