# Asterisk 1.8 Internal Call Drop out after 6 sec, wrong Contact **URL:** \ **Category:** Asterisk Support\ **Created:** [February 12, 2018, 6:03pm UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508 "2018-02-12T18:03:57Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![ruioke](https://avatars.discourse-cdn.com/v4/letter/r/ac8455/32.png) [@ruioke](https://community.asterisk.org/u/ruioke)\ **Post date:** [February 12, 2018, 6:03pm UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508/1 "2018-02-12T18:03:57Z")
Hi Guys, I have a really big problem with an Italian ISP . To connect a trunk SIP with it i use a public IP NATTED on my asterisk (NAT from Watchguard) All Works good until today, when i try a inbound call asterisk get it and it works ok for 6 second but after that the call is dropped. If i check with tshark it says that I have a Unrecognised SIP header. After that my ISP check log too and it says that my problem is on my Contact that show my internal ip and not my public ip. I check it too and i see the same issue ,this in the header: ![image](https://global.discourse-cdn.com/asterisk/original/2X/d/dbe35fd8eee62336490c6dc7ddfb01ee3105f32f.png) I’ve tried to override it with contact= in and check my externalip on sip.conf and it’s all ok. There is a way to change that parameter? These are my trunk settings: [e2abb71f-9263-49a0-9e5b-eff2f57c905e] ;%VoncSipProvider% username= secret= context=925bc138-4a77-40f6-bb51-e4798da69f02 host=ISP PUBLIC IP type=peer port=5060 qualify=yes insecure=invite,port canreinvite=yes dtmfmode=rfc2833 stunaddr= ;description=TRUNK SIP ;prefix= ;GenerateRing=1 accountcode=\ **Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 12, 2018, 6:56pm UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508/2 "2018-02-12T18:56:51Z") You probably have externip or localnets set wrongly, so Asterisk either doesn’t know that you have NAT, or thinks that it the ITSP is insider the NAT. ---
**Author:** ![ruioke](https://avatars.discourse-cdn.com/v4/letter/r/ac8455/32.png) [@ruioke](https://community.asterisk.org/u/ruioke)\ **Post date:** [February 13, 2018, 8:37am UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508/3 "2018-02-13T08:37:42Z")
I’m trying to get out of this issue but i don’t find nothing wrong in my sip.conf ```auto [general] port=5060 bindaddr=0.0.0.0 subscribecontext=BLF limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes nat=yes disallow=all maxexpiry=3600 defaultexpiry=3400 relaxdtmf=yes ;;MODIFICA DISABILITAZIONE JITTER BUFFER ;jbenable=yes ;jblog =yes ;jbforce=no ;jbenable=no ;jbmaxsize=160 ;jbresyncthreshold=160 ;jbimpl=adaptive ;usejb=no ;jbsize=600 ;jblog=yes language=it ; Default language setting for all users/peers ;jitterbuffers=50 videosupport=yes ; Turn on support for SIP video. You need to turn this on ;useragent=VoipNet ; Allows you to change the user agent string rtptimeout=60 rtpholdtimeout=300 ; incoming calls ;insecure = very ;register => 6998800:iifhbTmb@6998800.zeppelink.net:5060/9600 ; 0247951034 is the account and DID number for Italy t38pt_udptl= yes realm=VoipNetCallCenter callevents=yes Qualify=No externip= **MYPUBLICIP** /255.255.255.248 localnet=172.16.0.0/255.255.0.0 localnet=ISP PUBLICIP.0/255.255.255.0 localnet=93.42.189.86/255.255.255.255 localnet=192.168.0.35/255.255.255.0 rtcachefriends=yes allow=alaw allow=g729 allow=h261 allow=h264 allow=h263p allow=h263 allow=mpeg4 ``` I’ve tried to set netmask on my externip just in case but nothing change. > > ```auto > > eth0 Link encap:Ethernet HWaddr 00:50:56:91:2d:2b > > inet addr:172.16.14.100 Bcast:172.16.255.255 Mask:255.255.0.0 > > inet6 addr: fe80::250:56ff:fe91:2d2b/64 Scope:Link > > UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 > > RX packets:1480752 errors:0 dropped:177 overruns:0 frame:0 > > TX packets:1785464 errors:0 dropped:0 overruns:0 carrier:0 > > collisions:0 txqueuelen:1000 > > RX bytes:973498014 (973.4 MB) TX bytes:1085584053 (1.0 GB) > > > > ``` root@onc-callcenter-64:~# route Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface ```auto @PBX:~# route Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface host051.pool213 172.16.90.254 255.255.255.255 UGH 0 0 0 eth0 host054.pool213 172.16.90.254 255.255.255.255 UGH 0 0 0 eth0 **ISP_PUBLIC_IP** 172.16.90.254 255.255.255.0 UG 0 0 0 eth0 172.16.0.0 * 255.255.0.0 U 0 0 0 eth0 default 172.16.8.254 0.0.0.0 UG 100 0 0 eth0 ``` ---
**Author:** ![ruioke](https://avatars.discourse-cdn.com/v4/letter/r/ac8455/32.png) [@ruioke](https://community.asterisk.org/u/ruioke)\ **Post date:** [February 13, 2018, 9:08am UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508/4 "2018-02-13T09:08:34Z")
PBX-64\*CLI\> sip show settings ## Global Settings: UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: Yes Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: No Realm. auth: No Our auth realm VoipNetCallCenter Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.10.1 SDP Session Name: Asterisk PBX 1.8.10.1 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: On Auth. Failure Events: Off T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP\_CAUSE: No ## Network QoS Settings: IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No ## Network Settings: SIP address remapping: Enabled using externaddr Externhost: Externaddr: 123.123.126.54:0 (My Public IP) Externrefresh: 10 Localnet: 172.16.0.0/255.255.0.0 (LOCAL address pbx) 123.456.678.0/255.255.255.0 (ISP Public address) 93.42.189.86/255.255.255.255 192.168.0.0/255.255.255.0 ## Global Signalling Settings: Codecs: 0x7c0108 (alaw|g729|h261|h263|h263p|h264|mpeg4) Codec Order: alaw:20,g729:20 Relax DTMF: Yes RFC2833 Compensation: No Symmetric RTP: Yes Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 60 RTP Hold Timeout: 300 MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 3400 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: Yes SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 ## Default Settings: Allowed transports: UDP Outbound transport: UDP Context: default Force rport: Yes DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: it MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk * * * ---
**Author:** ![ruioke](https://avatars.discourse-cdn.com/v4/letter/r/ac8455/32.png) [@ruioke](https://community.asterisk.org/u/ruioke)\ **Post date:** [February 13, 2018, 9:32am UTC](https://community.asterisk.org/t/asterisk-1-8-internal-call-drop-out-after-6-sec-wrong-contact/73508/5 "2018-02-13T09:32:19Z")
Fixed it!! i’ve set the public ip of ISP on sip.conf localnet and for this reason asterisk think that it don’t need NAT. After remove localnet=ISPPUBLICIP all works. Hope this could help someone Fabio