search for: forcerport

...found to engine 'mysql', but the engine is not available [May 12 14:05:55] WARNING[2630] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available And on the asterisk console: kc3054*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 00000277611 (Unspecified) D N 0 Unmonitored j.witvliet (Unspecified) D N 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] [May 12 14:17:47] WARNING[2630]: config.c:2045 find_engine: Realtime map...

...figuring mysql-stuff: file: res_config_mysql.conf database access config: host, user, pwd file: res_odbc.conf in section [mysql2]: mysql database config: host, user, pwd So, i configured them both... Quick check:kc3054*CLI> sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheim default No Yes 00000277611 25b06d3a0b5ef73 default No Yes kc3054*CLI> kc3054*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Realtime 00000277611 (Unspecified) D N 0 Unmonitored j.witvlie...

...peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime 303411/303411 172.16.1.12 D 5060 OK (77 ms) Cached RT 467577/467577 172.16.1.22 D 5060 OK (141 ms)...

...RealTime Configuration Driver 0 > app_mysql.so Simple Mysql Interface 0 > 3 modules loaded > kc3054*CLI> > kc3054*CLI> sip show users > Username Secret Accountcode Def.Context ACL ForcerPort > j.witvliet geheim default No Yes > 00000277611 25b06d3a0b5ef73 default No Yes > kc3054*CLI> > kc3054*CLI> sip show peers > Name/username Host...

I recently received a GoIP-32 for a client project -- primarily outbound calling. How should a GoIP be configured for Asterisk? No fancy shmancy Elastix or FPBX GUI -- just using the configuration files. Single Server Mode, Config By Line, and Trunk Gateway Mode all seem likely suspects. How did you configure your GoIP and why? What do your relevant sip.conf section(s) look like? What does

...a call to user 11 from twinkle and I get the IVR message for 'the number you have called is not in service'. I have tried to register Twinkle and this always fails. If I do : # asterisk -rvvvvvvvv CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] CLI> sip show users Username Secret Accountcode Def.Context ACL ForcerPort Which seems to say to me that I have nothing configured albeit I h...

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) My question is how can we get Asterisk to fill...

...SIP trunk connected to it. I checked and noted that sip show peers in the CLI would either do nothing (e. g. just show asterisk*cli> again, with no response) or it would return only this: asterisk*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status asterisk*CLI> A module show like sip also does literally nothing, just asterisk*cli> module show like sip asterisk*CLI> Soon after this, I lost the capacity to get any response if I do an asterisk -r on the commandline - it would just hang indefintely. Did a reboot,...

...as expected. Where I run into problems is in Inbound calls. When I try to call the extension from a DID I am receiving "Unable to create channel of type 'SIP' (cause 20 - Unknown)". And rightfully so! Reason being: SIP Show Peers Yields: Name/username Host Dyn Forcerport ACL Port Status Realtime 1001/1001 192.168.2.5 N 5060 UNREACHABLE Cached RT TTrunk/sip.exp.com 192.168.2.5 N 5060 UNKNOWN Cached RT As for who will keep track of the UA location, the OpenSIPS `location` table has the correct info: se...

...dynamic canreinvite=no ; (dtmf not wroking correctly without this one) disallow=all allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup= My asterisk registration is correct as well: sip show users Username Secret Accountcode Def.Context ACL Forcerport pstn-9998 158567 incoming No No pstn-4444 spa354 incoming No No Caller display ID from PSTN on FXO ports are working OK. The [pstn-4444] is channel: 4 The [pstn-9998] is channel: 3 If th...

...show the same issue when request show peers. Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199) PBX*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 100/199 192.168.11.160 D Yes Yes A 5060 OK (30 ms) I check the sip 100 and (aparently) show all normal PBX*CLI> sip show peer 100 * Name : 100 Description : Secret...

...s://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+stru cture> https://wiki.asterisk.org/wiki/display/ ... +structure) The user registered on asterisk works fine, but not the peer. I'd like to use my voipdiscount account as a peer to do external call. Name/username Host Dyn Forcerport ACL Port Status Realtime 2000/2000 xxx.xxx.xxx.xxx D N 65476 OK (117 ms) Cached RT Mysql entry on sipclient table is below: "3" "XXXXXXXX "sip.voipdiscount.com" "5060" \N "XXXXXXXXXX" \N \N \N \N "sip.voipdiscount.com" "peer" &quo...

...d non-zero > on 'SIP/6000-00000013' in macro 'stdexten' > == Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on > 'SIP/6000-00000013' > *CLI> sip show peers > Name/username Host Dyn > Forcerport ACL Port Status > * 123456789101112/6201 > 192.168.0.102 N 5060 > Unmonitored* > 6000/6000 192.168.0.102 > D N 5061 Unmonitored > 6001/6001 192.168.0.102 > D N...

...189 handle_request_invite: Failed to authenticate device "Ext 110" < sip:110 at 192.168.6.2>;tag=1130259112 Asterisk 192.168.6.2 OpenVPN on router 10.8.0.1 Remote Yealink phone 10.8.0.6 The remote phone shows as being registered: PBX*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN Also, if there is voicemail in the mailbox for 110, the phone's message light is lit and it beeps periodically. toshi*CLI> sip show peer 110 * Name : 110 Description : Yealink Op...

...at context > in extensions.conf. Mind you, then 200 is the only phone that can dial > out. 201 can only dial 200 and nothing else. Wait a minute, slow down. I re-installed, same sort of problem: vici:~ # vici:~ # asterisk -rx "sip show peers" Name/username Host Dyn Forcerport ACL Port Status 300/300 (Unspecified) D N 0 UNKNOWN 301/301 192.168.0.24 D N 5060 OK (29 ms) 302/302 (Unspecified) D N 0 UNKNOWN gs102/gs102 (Unspecified) D N...

...ination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 Well, the phone connect to the server and I can see it reachable: OpenWrt*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 00491771111111/0049177111 192.168.200.3 D N 40702 OK (1768 ms) Well, now I call the mobile phone from another peer. It rings and I can answer the call. Wonderful! But no word will be sent... :( I cannot hear anything on my mobil...

...; 102,1,Dial(SIP/celular) exten => 103,1,Dial(SIP/fxs) exten => 104,1,Answer() same => n,Playback(tt-weasels) same => n,Hangup() include => outgoing-call This is the result of "sip show peers" -------------------------------------- Name/username Host Dyn Forcerport Comedia ACL Port Status Description celular/celular 192.168.1.21 D Yes Yes 47747 OK (6 ms) samsung gt-s7562 fxs/fxs 192.168.1.4 D Yes Yes 5060 OK (27 ms) fxs port1 pstn-out 192.168.1.4 No No...

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123 password 123 default No Yes devries password devries default No Yes babytel...

...: Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No tleilax*CLI> tleilax*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 201/201 (Unspecified) D N 0 UNKNOWN babytel/1<private> 198.38.7.11 D N 5060 UNREACHABLE gs102/gs102 (Unspecified) D N 0 UNKNOWN 3 sip peers [Monitored: 0 onli...

...lly understand what's going on here, neither how it works now, nor what I need to do to make Avaya's SM happy. * When I do sip show peers on my IVR box, I see the Avaya session manager: jerec*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status sessionmgr1 10.10.0.17 5060 OK (1 ms) * The Avaya engineer says he is seeing "SIP/2.0 400 Bad FROM header" in his trace screen, and his SM status screen shows "500 NOT REACHABLE" as the status...