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
