[asterisk-users] Asterisk trunking between two location

Gopalakrishnan N gopalakrishnan.an at gmail.com
Tue Jul 2 15:56:10 CDT 2013


I tried creating two trunks with following,
*1st Location*
[10.30.2.5]
type=friend
username=indman01
secret=indman01
host=dynamic
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

*2nd Location*
[10.20.111.48]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=india
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

My dialplan is like this
exten => _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN})
exten => _2XXX,n,Hangup

And the output I get is
 Executing [2001 at Test:1] Dial("SIP/3081-000027d2", "SIP/10.30.2.5/2001") in
new stack
[Jul  2 16:49:57] WARNING[15766][C-00002b94]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [2001 at Test:2] Hangup("SIP/3081-000027d2", "") in new stack
  == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-000027d2'

Actually the trunk which i mentioned in my first email, it was working...
and from today it is not....

Still breaking... what could be the reason... !



On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad <asghar144 at gmail.com> wrote:

> yes you can. just create trunks on both side with static ip and in dial
> use trunk name.
> exten => _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =>
> _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
> make a call from a to b and one from b to and post cli log here or upload
> anyware else.
>
>
> On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> can't we use without register command both way as peer to peer?
>>
>>
>> On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>
>>> 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
>>> and 10.10.10.0 on a.
>>> 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
>>> B.
>>> 3. general section in sip.conf of side B register with server A.
>>>
>>> please see comments in sip.conf
>>> ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
>>> registering
>>>                                 ; as any IP address used for staticly
>>> defined
>>>                                 ; hosts.  This helps avoid the
>>> configuration
>>>                                 ; error of allowing your users to
>>> register at
>>>                                 ; the same address as a SIP provider.
>>>
>>>
>>>
>>> On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N <
>>> gopalakrishnan.an at gmail.com> wrote:
>>>
>>>> [servera]
>>>> type=friend
>>>> username=servera
>>>> secret=servera
>>>> host=10.30.2.5
>>>> port=5060
>>>> context=Manila
>>>> insecure=port,invite
>>>> dtmfmode=rfc2833
>>>> relaxdtmf=yes
>>>> directmedia=no
>>>> qualify=yes
>>>> disallow=all
>>>> allow=g729
>>>> allow=ulaw
>>>> allow=alaw
>>>> deny=0.0.0.0/0.0.0.0
>>>> permit=10.30.2.5/255.255.255.0
>>>>
>>>> If i use host=dynamic, it wont communicate each other and will result
>>>> to unmonitored....
>>>>
>>>>
>>>> and the IP segment is two different segment. where am able to ping each
>>>> other.
>>>>
>>>>
>>>>
>>>> On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>>>
>>>>> hi,
>>>>> paste server a trunk also, if you want register why you are not using
>>>>> host=dynamic?
>>>>> both servers are on 10.10.10.0 ? if no then check your deny permit
>>>>> seting.
>>>>>
>>>>>
>>>>> On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N <
>>>>> gopalakrishnan.an at gmail.com> wrote:
>>>>>
>>>>>> Also tried one more scenario, particularly from one IP to other IP
>>>>>> not registering.
>>>>>>
>>>>>> For example like 10.10.10.5 to 10.20.10.5
>>>>>>
>>>>>> If it is 10.10.10.5 to 10.30.2.5 - working
>>>>>> If it is 10.30.2.5 to 10.20.10.4 works fine.
>>>>>>
>>>>>> really strange... I suspect some issue on the network side...
>>>>>>
>>>>>> Problem is there is no packet loss.. with mtr it is fine, tracepath
>>>>>> is fine, ping is fine... :(
>>>>>>
>>>>>>
>>>>>> On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N <
>>>>>> gopalakrishnan.an at gmail.com> wrote:
>>>>>>
>>>>>>> Am using Asterisk 11.2 in one location and 11.1 in another location.
>>>>>>>
>>>>>>> when I trunk between two servers, the status is unreachable.
>>>>>>>
>>>>>>> But with different server with 11.2 and 11.2 it works fine.
>>>>>>>
>>>>>>> I tried both IAX and SIP.
>>>>>>>
>>>>>>> the trunk in sip.conf what i have is,
>>>>>>> [serverb]
>>>>>>> type=friend
>>>>>>> username=serverb
>>>>>>> secret=serverb
>>>>>>> host=10.10.10.5
>>>>>>> port=5060
>>>>>>> context=default
>>>>>>> insecure=port,invite
>>>>>>> dtmfmode=rfc2833
>>>>>>> relaxdtmf=yes
>>>>>>> directmedia=no
>>>>>>> qualify=3000
>>>>>>> nat=force_rport,comedia
>>>>>>> disallow=all
>>>>>>> allow=g729
>>>>>>> allow=ulaw
>>>>>>> allow=alaw
>>>>>>> deny=0.0.0.0/0.0.0.0
>>>>>>> permit=10.10.10.5/255.255.255.0
>>>>>>>
>>>>>>> Is there any issue with 11.1?
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>>                http://www.asterisk.org/hello
>>>>>>
>>>>>> asterisk-users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>                http://www.asterisk.org/hello
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>                http://www.asterisk.org/hello
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>                http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130703/bd661913/attachment.htm>


More information about the asterisk-users mailing list