[asterisk-users] Asterisk trunking between two location

Gopalakrishnan N gopalakrishnan.an at gmail.com
Tue Jul 2 20:23:16 CDT 2013 

By having different server, i made it work. I suspect some network issue...


On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad <asghar144 at gmail.com> wrote:

> make a call and post cli log
>
>
> On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> still the peer shows unreachable.... let me restart and give a try...
>>
>>
>> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>
>>> *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
>>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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/c96ff3f0/attachment.htm>

More information about the asterisk-users mailing list