By having different server, i made it work. I suspect some network issue...
On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad <[email protected]> wrote: > make a call and post cli log > > > On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N < > [email protected]> wrote: > >> still the peer shows unreachable.... let me restart and give a try... >> >> >> On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad <[email protected]>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 < >>> [email protected]> 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@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@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 <[email protected]>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 < >>>>> [email protected]> 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 >>>>>> <[email protected]>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 < >>>>>>> [email protected]> 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 < >>>>>>>> [email protected]> 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 < >>>>>>>>> [email protected]> 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 < >>>>>>>>>> [email protected]> 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 >
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