SIP Dialing Issues

Asterisk Asterisk Support
1

Hi again everyone!

Thanks to my lack knowledge and general incompetence, I’m still sat here literally pulling my hair out with this problem. Hoping someone could point out my configuration problem… if you could I’ll love you forever more :wink:

Asterisk V1.4.18, pretty much out of the box configuration.

Added registration line into ‘sip.conf’ which appears to have done the trick:

register => 123456:mypassword@sip.gradwell.net/5000

(For the record, yes, in reality the correct username and password has been entered).

Incoming calls work fine. But as the title suggests, outgoing calls don’t work.

‘users.conf’ has the following:

[quote][5000]
fullname = Colin
email = thisis@myemail.com
secret = 1234
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = numberplan-custom-1
cid_number = 123456789
hasagent = yes
hasdirectory = no
deletevoicemail = no
host = dynamic
mailbox = 5000
threewaycalling = no
registeriax = no
registersip = yes
canreinvite = no
nat = yes
dtmfmode = rfc2833
allow = ulaw,alaw,g726

[Gradwell]
allow = ulaw,gsm,g729,g726
context = DID_Gradwell
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = sip.gradwell.net
port = 5060
registeriax = no
registersip = yes
secret = mypassword
trunkname = Custom - Gradwell
trunkstyle = customvoip
username = 123456
callerid = 123456789
fromdomain = sip.gradwell.net
fromuser = 123456
insecure = very
disallow = alaw,ilbc,speex,adpcm,lpc10
[/quote]

And in ‘extensions.conf’ I’ve got:

[quote][numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten = _0X!,1,Macro(trunkdial,${Gradwell}/${EXTEN:0})
comment = _0X!,1,Gradwell,standard

[DID_Gradwell]
include = default
include = default
exten = _123456789,1,Goto(default|5000|1)
[/quote]

Everytime I attempt to make a call, I’ve presented with:

Most of the configuration was generated by the Asterisk GUI. Sorry to stick all this code in here, but any pointers would be helpful. Thanks for your time in advance!

Colin

2

It looks like that you want to make outgoing call but failed. please test that account with x-lite first. if x-lite can call out, then turn to debug the asterisk.

It is better to post your asterisk kernal output to debug.

such as “asterisk -rvvvvv”

telecomchinasourcing.com/knowledge

3

I have the same problem.
This looks like bug.
When I configure user in users.conf it is in default context despite of I set different context in users.conf

sip show peer 300

  • Name : 300
    Secret :
    MD5Secret :
    Context : numberplan-custom-1
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup : 1
    Pickupgroup : 1
    Mailbox : 300
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 0
    Dynamic : Yes
    Callerid : “test” <300>
    MaxCallBR : 384 kbps
    Expire : 2782
    Insecure : no
    Nat : RFC3581
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : rfc2833
    LastMsg : 0

but dialplan shows something strange

dialplan show default
[ Context ‘default’ created by ‘pbx_config’ ]
‘1111’ => 1. Answer() [pbx_config]
2. Playback(hello-world) [pbx_config]
3. Hangup() [pbx_config]
‘300’ => hint: SIP/300&IAX2/300 [pbx_config]
1. Dial(${HINT}) [pbx_config]

anyone solved this issue with gui ?
when I add user to sip.conf it is in right context.

Regards,
Arek

4

exten = _0X!,1,Macro(trunkdial,${Gradwell}/${EXTEN:0})

Is _0X! what you want?

What happens when you use _X!?