Problem with calling Out

Asterisk Asterisk Support
1

Hi, I tried to configure my sip provider with the web interface, I check the user.conf, and tried everyting and nothig work. I check if my provider register me and the sip show registry command told me that I’m registered. Here is the error I got when I tried to place an outside call:

– Executing [(THE NUMBER I DIAL)@numberplan-custom-1:1] Macro(“SIP/227-08210278”, "trunkdial|SIP/trunk_1/( The number I dial )in new stack
– Executing [s@macro-trunkdial:1] Dial(“SIP/227-08210278”, “SIP/trunk_1/7874350473”) in new stack
– Called trunk_1/7874350473
[Mar 23 07:26:29] WARNING[5367]: chan_sip.c:11731 handle_response_invite: Received response: “Forbidden” from ‘“Comp” <sip:(MY SIP PHONE NUMBER)@10.5.148.25>;tag=as5c282715’
– SIP/trunk_1-08213288 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-trunkdial:2] Goto(“SIP/227-08210278”, “s-CONGESTION|1”) in new stack
– Goto (macro-trunkdial,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-trunkdial:1] NoOp(“SIP/227-08210278”, “”) in new stack

My trunk in the user.conf is this

[trunk_1]
secret = (MY PASSWORD)
provider =
trunkstyle = customvoip
username = (MY USER NAME)
trunkname = Custom - MY PROVIDER NAME
callerid = (MY SIP PHONE NUMBER)
hasexten = no
hassip = yes
hasiax = no
registeriax = no
registersip = yes
host = (MY SIP PROVIDER)
dialformat = ${EXTEN:1}
context = DID_trunk_1
group =
insecure =
fromuser = (MY SIP PHONE NUMBER)
fromdomain =

With my other provider(Stanaphone) work perfectly, I don’t known why this new one don’t work.

Any idea?

2

Hi to all, now I can place an outside call, my regular phone ring and I answer the phone but I can’t hear nothing after few seconds the call is dropped. I don’t know if my problem is the type of codec i use.

This is part of my sip configuration for my sip provider;

[SIP PROVIDER]
type = peer
;user = phone
host = MY SIP PROVIDER
fromdomain = MY SIP PROVIDER
fromuser = MY SIP PHONE
secret= MY PASSWORD
username= MY USERNAME
insecure = very
context = MY PROVIDER
authname = MY USERNANE
dtmfmode = inband
dtmf = inband
;Disable canreinvite if you are behind a NAT
canreinvite = no

Any Idea ?

3

Try playing around with the NAT setting. Makes a lot of difference for some phone models.

Rgds

Patrick Arkley
www.say-no.se

4

Thanks for the answer, the problem was the Nat, now is working

5

Hi Arosa

Please could you explain how do you resolved the problem for the outgoing calls? I have the same problem and i can’t resolve it
thanks

6

I am also having this same problem… any post you could give would be appreciated.

7

NAT issues are covered pretty extensively on the wiki at voip-info.org and on other pages (search google for “asterisk nat”) or if you search this forum you’ll find the same question asked and answered many many times.

8

Are you indicating a nat issue between the phone and the A box or are you saying between the A box and the sip provider?

9

it doesn’t really matter … if there’s a NAT between Asterisk and a peer or user then this needs to be accounted for in the configs.

10

Well, it does matter.

I had 2 Polycom 501 and 4 Cisco 7940 on a private network. The Asterisk where also on the same lan.
Regardless of having NAT set to yes or no for the Polycom peers the Polycoms could register. The Ciscos couldn’t. When I set the Cisco peers to NAT=no they could register as well.

I’m not sure thid is covered on the NAT section.

Rgds

Patrick Arkley
www.say-no.se

11
12

Sorry, was a bit trigger happy :wink:

/Patrick Arkley (ww.say-no.se)

13

guys i’m not talking about the NAT problem, i’m talking about the Arosa’s first problem, so the post. (CONGESTION)
:smiley:

14

this is what the OP actually said

15

Hi to all in the forum, to resolve the problem I got, the first thing I did was to place the Asterisk Server in the DMZ zone in the router to eliminate some NAT problem, then I create my sip provider with the web interface. Then I modify the user.conf, where the sip provider is created. The interesting part is when I create my sip provider with the web interface I need to fill some settings, like the (provider = My Provider , insecure = very, fromdomain=My Provider). Then when I test and everything is working I remove the server from the DMZ and configure the ports the asterisk server needs, (UDP Ports 5060, 10000-20000)

[trunk_1]
secret = (MY PASSWORD)
provider =
trunkstyle = customvoip
username = (MY USER NAME)
trunkname = Custom - MY PROVIDER NAME
callerid = (MY SIP PHONE NUMBER)
hasexten = no
hassip = yes
hasiax = no
registeriax = no
registersip = yes
host = (MY SIP PROVIDER)
dialformat = ${EXTEN:1}
context = DID_trunk_1
group =
insecure =
fromuser = (MY SIP PHONE NUMBER)
fromdomain =

16

Thanks Guys but i still have the problem… :imp: :imp: :frowning: :frowning:
I don’t know what else to try.