Current configuration follows: [general] context=default allowguest=no alwaysauthreject=yes allowoverlap=yes allowtransfer=yes tcpenable=no tlsenable=no srvlookup=yes vmexten=vm rtcachefriends=yes nat=no directmedia=nonat directrtpsetup=no videosupport=yes maxcallbitrate=384 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm allow=ilbc allow=speex allow=g726 allow=g723 mohinterpret=default mohsuggest=default dtmfmode=rfc2833 timer1b=60000 transport=udp
[carrier-1] host=a.b.c.d type=friend context=from-pstn disallow=all allow=ulaw,alaw qualify=yes trunk=yes [90102] secret=xxxxxx mailbox=90102@default cid_number=NXXXXXXXXX accountcode=401 type=friend host=dynamic port=5060 qualify=yes nat=yes transport=udp context=users disallow=all allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263 directmedia=no canreinvite=no videosupport=no On Wed, May 15, 2013 at 2:47 PM, Asghar Mohammad <[email protected]>wrote: > please show us peer configuration. > > > On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk > <[email protected]>wrote: > >> Users (softphones) are behind a NAT, Asterisk has its own public ip >> address >> >> >> On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad <[email protected]>wrote: >> >>> asterisk is behind nat? >>> >>> >>> On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk <[email protected] >>> > wrote: >>> >>>> Hello everyone, >>>> >>>> I've suffering cut offs after 6 or 7 seconds a call is answered, >>>> incoming calls are working fine, but outgoing ones show the gollowing >>>> messages when are being dropped: >>>> >>>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: >>>> Retransmission timeout reached on transmission >>>> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical >>>> Response) -- See >>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions >>>> Packet timed out after 6399ms with no response >>>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: >>>> Hanging up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to >>>> our critical packet (see >>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). >>>> This is happening with my PBX hosted on an external network and peers >>>> on my local network. >>>> >>>> It seems the SIP ACK is not being received properly. >>>> >>>> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9 >>>> >>>> Elder D. Arohuanca >>>> Lima - Peru >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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
