Topic: SAIL - no outbound calls

[frifri]

Hi,

I have installed SAIL 2.1.14-414 in Server-Gateway mode on SME 7.1, and X-Lite as extension 5000.
The extension registrated correctly.

Trunkconfig is like this :
type=peer
host=ipness.net
port=6060
qualify=3000
canreinvite=no
username=pollet
fromuser=pollet
secret=********
disallow=all
allow=alaw
allow=ulaw
allow=g729
insecure=very

Initialisationstring is like this :
DID:********:username@ipness.net:6060

I have one route : _XXXXXXXX.

I can make inbound calls, but no outbound calls.  X-Lite says that the outbound call is established, but the other end doesn't ring. There is un 'hang up' after 10 sec.

Please help

[hervep]

Hi,

I have installed SAIL 2.1.14-414 in Server-Gateway mode on SME 7.1, and X-Lite as extension 5000.
The extension registrated correctly.

Trunkconfig is like this :
type=peer
host=ipness.net
port=6060
qualify=3000
canreinvite=no
username=pollet
fromuser=pollet
secret=********
disallow=all
allow=alaw
allow=ulaw
allow=g729
insecure=very

Initialisationstring is like this :
DID:********:username@ipness.net:6060

I have one route : _XXXXXXXX.

I can make inbound calls, but no outbound calls.  X-Lite says that the outbound call is established, but the other end doesn't ring. There is un 'hang up' after 10 sec.

Please help

Hi,

Might be a codec problem ...
As far as I know, X-lite is not G729 compliant ( free version ? ).
I don't know ipness.net, but I guess they only accept G729.

Easy test is to use some hardware set instead of 'x-lite'. Most of them have G729 support.

Regards,

Herve

[frifri]

Don't think its the codec, because i can use X-Lite and Ipness without Asterisk/Sail.  Only when i set the Asterisk/Sail in front, i can't make outbound calls ...  I have never problems with incomming calls.

[hervep]

Don't think its the codec, because i can use X-Lite and Ipness without Asterisk/Sail.  Only when i set the Asterisk/Sail in front, i can't make outbound calls ...  I have never problems with incomming calls.

Let's look further then ...

* Did you ALLOW external calls in your extension definition ?
=> extensions / edit / Call Permissions [External]

* Your route definition sounds strange to me. I assume you are calling something like 071/123456 ( Belgium ? ). let's define the route (first choice = trunk to provider) using '_0.' instead of '_XXXXXXX' . Try a call without access code ( Not 0071123456 but 071123456 ).

Hervé

[frifri]

Hi Hervé,

Its indeed for belgian numbers (012/345678).  I 've changed the route like you explained, but no joy ...

Extension is set up like this :
type=friend
username=5000
secret=5000
host=dynamic
qualify=3000
context=internal
callerid="5000" <5000>
canreinvite=no
mailbox=5000
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw

Call permissions is set to external.

I hope you can find the problem, because i really want to have this working.

[hervep]

OK, let's continue ...

* If you call an internal extention or voicemail ... it works ( Pls confirm )

* Pls remove 'allow=g729' on your trunk.

* If it's still not working, pls make a CLI trace of a call attempt ...

Hervé

[frifri]

* i can call from outsite to extensions insite, and i can call from one extension to an other (i also installed IDEFisk (AIX2)).
* i removed 'allow=g729' on the trunk.
* how can i make a CLItrace of a call attempt ?
* could it be a DNS-problem ?

[hervep]

* i can call from outsite to extensions insite, and i can call from one extension to an other (i also installed IDEFisk (AIX2)).
* i removed 'allow=g729' on the trunk.
* how can i make a CLItrace of a call attempt ?
* could it be a DNS-problem ?

To make a cli trace :
- you need to login into your system as root
- at the command prompt  

Code: [Select]

asterisk -vvvvvvvvvvvvvvvvvr
- you will get something like CLI>

Make a call attempt, and look at the CLI.

DNS : I don't think so, otherwise you should also have problems using 'x-lite' directly connected to 'ipness.net' as you mentionned before.

Hervé

[frifri]

This is what i get after the CLItrace :

Verbosity was 0 and is now 27
    -- Executing AGI("SIP/5000-0844d798", "selintra|OutCluster|051611311") in ne                                                                                             w stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing AGI("SIP/5000-0844d798", "selintra|OutRoute|Normaal") in new st                                                                                             ack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (SetCallerID) Options: (3292120333)
    -- AGI Script Executing Application: (Dial) Options: (SIP/051611311@32921203                                                                                             33)
    -- Called 051611311@3292120333
Feb 23 08:49:02 WARNING[10478]: chan_sip.c:9694 handle_response_invite: Forbidde                                                                                             n - wrong password on authentication for INVITE to '"3292120333" <sip:pollet@84.                                                                                             198.134.155>;tag=as74953a88'
    -- SIP/3292120333-08453300 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
Feb 23 08:49:02 WARNING[13858]: file.c:512 ast_openstream_full: File were-sorry                                                                                              does not exist in any format
Feb 23 08:49:02 WARNING[13858]: file.c:824 ast_streamfile: Unable to open were-s                                                                                             orry (format alaw): No such file or directory
Feb 23 08:49:02 WARNING[13858]: pbx.c:5798 pbx_builtin_background: ast_streamfil                                                                                             e failed on SIP/5000-0844d798 for were-sorry
    -- AGI Script Executing Application: (Background) Options: (call-cannot-comp                                                                                             lete)
Feb 23 08:49:02 WARNING[13858]: file.c:512 ast_openstream_full: File call-cannot                                                                                             -complete does not exist in any format
Feb 23 08:49:02 WARNING[13858]: file.c:824 ast_streamfile: Unable to open call-c                                                                                             annot-complete (format alaw): No such file or directory
Feb 23 08:49:02 WARNING[13858]: pbx.c:5798 pbx_builtin_background: ast_streamfil                                                                                             e failed on SIP/5000-0844d798 for call-cannot-complete
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-a                                                                                             nd-try-again)
Feb 23 08:49:02 WARNING[13858]: file.c:512 ast_openstream_full: File please-hang                                                                                             -up-and-try-again does not exist in any format
Feb 23 08:49:02 WARNING[13858]: file.c:824 ast_streamfile: Unable to open please                                                                                             -hang-up-and-try-again (format alaw): No such file or directory
Feb 23 08:49:02 WARNING[13858]: pbx.c:5798 pbx_builtin_background: ast_streamfil                                                                                             e failed on SIP/5000-0844d798 for please-hang-up-and-try-again
    -- AGI Script selintra completed, returning 0
    -- Timeout on SIP/5000-0844d798
  == CDR updated on SIP/5000-0844d798
    -- Executing Hangup("SIP/5000-0844d798", "") in new stack
  == Spawn extension (default, t, 1) exited non-zero on 'SIP/5000-0844d798'
    -- Executing Hangup("SIP/5000-0844d798", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5000-0844d798'

[hervep]

Hi frifri,

OK things becomes more clear ...

1) You get the feeling that you are connected just because you don't hear feedback from your system. Without going into details, I think you did not
load the sounds rpm.

Pls proceed with the following using rpm -Uvh :

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/smeserver-asterisk-sounds-1.2.2-2.noarch.rpm

2) Looks to be some authentication issue. user/password does not seems to be accepted. Pls control your account data.

Code: [Select]

Feb 23 08:49:02 WARNING[10478]: chan_sip.c:9694 handle_response_invite: Forbidde n - wrong password on authentication for INVITE to '"3292120333" <sip:pollet@84. 198.134.155>;tag=as74953a88'


You can go further into the trace by making a new test, with sip debug enabled :

Asterisk -vvvvvvvvvvvvvvvvvvvvr
CLI> sip debug

Make test call ...

CLI> sip no debug

Best,

Hervé

[hervep]

Was looking further ...

Try with 'fromuser=3292120333' and 'username=3292120333' (assuming is is your nr) instead of pollet ...

Trunkconfig is like this :
type=peer
host=ipness.net
port=6060
qualify=3000
canreinvite=no
username=pollet
fromuser=pollet
secret=********
disallow=all
allow=alaw
allow=ulaw
insecure=very

best,

Hervé

[frifri]

Hi Herve,

Got it working, with this configuration for IPNESS :

Registration String : 32XXXXXXXX:password:username@ipness.net:6060

Sip.conf :

[32XXXXXXXX]
type=peer
host=ipness.net
fromdomain=ipness.net
port=6060
username=<username>
fromuser=<32XXXXXXXX>
secret=<password>
insecure=very
qualify=3000
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=g729

Perhaps Selintra can make a carrier-template of it.

Thanks for your great help !

[hervep]

Hi frifri,

Nice !

Tnx for the feedback.

Hervé