[asterisk-users] Asterisk trunking between two location

Asghar Mohammad asghar144 at gmail.com
Tue Jul 2 16:19:41 CDT 2013


*1st Location*
[manila]
type=peer
username=indman01
secret=indman01
host=10.30.2.5 <-- ip of 2nd location
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

1st location dialplan
exten => _2XXX,1,Dial(SIP/manila/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>)
exten => _2XXX,n,Hangup

*2nd Location*
[india]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=10.20.111.48 <- ip of 1st location
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

2st location dialplan
exten => _2XXX,1,Dial(SIP/india/${EXTEN} <http://10.30.2.5/$%7BEXTEN%7D>)
exten => _2XXX,n,Hangup

then you should handle the call when it arrive in any server
let me know if it work.


On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N <
gopalakrishnan.an at gmail.com> wrote:

> 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/${EXTEN}<http://10.30.2.5/$%7BEXTEN%7D>
> )
> 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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/20130702/ef8ef791/attachment.htm>


More information about the asterisk-users mailing list