make a call and post cli log
On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N < gopalakrishnan...@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 <asghar...@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...@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@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 <asghar...@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...@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 >>>>> <asghar...@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...@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 <asghar...@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...@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...@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