SIP Trunk

Hi

I’m a newby on the Asterix/Trixbox system.

we have just migrated office and we have moved from a legacy land line to a SIP architecture on which I’m having some issues to configure the trunks to work properly.

i have followed the procedures to add the truck as well as configure the incomming/outgoing routes but im still unable to dial in or out.

is there any advise anyone might be able to provide?

please let me know what info i need to post if any.

again, sorry but im quite a newby on this.

regards

1 - Perhaps you may have seen this. It is applicable to you also:
http://www.freepbx.org/forum/general-help/read-before-posting-help-us-help-you

2 - Did you install trixbox for this deployment? trixbox has been dead for years

hi

this is a existing installation that im trying to migrate to use SIP trunks as we dont have access to the landline connection on which it previously worked.

Version: 2.6.2.5
PBX: Asterix

Observation - it does though look like there is activity coming in.

any help would be greatly appreciated.

hi

my incoming is working but the outgoing is not.

kind regards

Good Morning.

We still do not know what you are using.

There is no such thing as Asterix (as far as I am aware)

To help us help you lets see if we can get away with a call trace.
Open up your console:
ssh in and run 'asterisk -rvvvvddd’
Make an outbound call.
Post the output here between tags
remove any sensitive information such as phone numbers. Replace them with X’s
Example 5558675309 becomes XXXXXXXXXX

Thanks

Hi

Find below the results


[trixbox.tagsa.co.za ~]# asterisk -rvvvvddd
Asterisk 1.6.0.26-FONCORE-r78, Copyright © 1999 - 2010 Digium, Inc. and others .
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: Parsing /etc/asterisk/asterisk.conf
== Found
== Parsing ‘/etc/asterisk/extconfig.conf’: Parsing /etc/asterisk/extconfig.con f
== Found
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox (pid = 2 277)
Verbosity was 3 and is now 4
Core debug was 0 and is now 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Extension Changed 103[ext-local] new state InUse for Notify User 117
== Extension Changed 103[ext-local] new state InUse for Notify User 103
== Extension Changed 103[ext-local] new state InUse for Notify User 100
== Extension Changed 103[ext-local] new state InUse for Notify User 110
== Extension Changed 103[ext-local] new state InUse for Notify User 101
– Executing [[email protected]:1] Goto(“SIP/103-00000000”, “ext-group,600,1”) in new stack
– Goto (ext-group,600,1)
– Executing [[email protected]:1] Macro(“SIP/103-00000000”, “user-callerid,”) i n new stack
– Executing [[email protected]:1] Set(“SIP/103-00000000”, “AMPUSER=103” ) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/103-00000000”, “0?report” ) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/103-00000000”, “1?Set(REA LCALLERIDNUM=103)”) in new stack
– Executing [[email protected]:4] Set(“SIP/103-00000000”, “AMPUSER=103” ) in new stack
– Executing [[email protected]:5] Set(“SIP/103-00000000”, “AMPUSERCIDNA ME=Anneline”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/103-00000000”, “0?report” ) in new stack
– Executing [[email protected]:7] Set(“SIP/103-00000000”, “AMPUSERCID=1 03”) in new stack
– Executing [[email protected]:8] Set(“SIP/103-00000000”, “CALLERID(all )=“Anneline” <103>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/103-00000000”, “0?Set(CHA NNEL(language)=)”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/103-00000000”, “0?contin ue”) in new stack
– Executing [[email protected]:11] Set(“SIP/103-00000000”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/103-00000000”, “1?contin ue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/103-00000000”, “Using Call erID “Anneline” <103>”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/103-00000000”, “1?skipdb”) in new stack
– Goto (ext-group,600,4)
– Executing [[email protected]:4] Set(“SIP/103-00000000”, “__NODEST=”) in new s tack
– Executing [[email protected]:5] Set(“SIP/103-00000000”, “__BLKVM_OVERRIDE=BLK VM/600/SIP/103-00000000”) in new stack
– Executing [[email protected]:6] Set(“SIP/103-00000000”, “__BLKVM_BASE=600”) i n new stack
– Executing [[email protected]:7] Set(“SIP/103-00000000”, “DB(BLKVM/600/SIP/103 -00000000)=TRUE”) in new stack
– Executing [[email protected]:8] Set(“SIP/103-00000000”, “RRNODEST=”) in new s tack
– Executing [[email protected]:9] Set(“SIP/103-00000000”, “__NODEST=600”) in ne w stack
– Executing [[email protected]:10] Set(“SIP/103-00000000”, “RecordMethod=Group” ) in new stack
– Executing [[email protected]:11] Macro(“SIP/103-00000000”, “record-enable,100 ,Group”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/103-00000000”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/103-00000000”, “recordingche ck,20140521-171134,1400685094.0”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
– <SIP/103-00000000>AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/103-00000000”, “”) in new stack
– Executing [[email protected]:12] Set(“SIP/103-00000000”, “RingGroupMethod=rin gall”) in new stack
– Executing [[email protected]:13] Macro(“SIP/103-00000000”, "dial,20,TtrwW,100 ") in new stack
– Executing [[email protected]:1] GotoIf(“SIP/103-00000000”, “1?dial”) in new st ack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/103-00000000”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘xxxxxx’ number is '103’
dialparties.agi: Methodology of ring is ‘ringall’
– dialparties.agi: Added extension 100 to extension map
– dialparties.agi: Extension 100 cf is disabled
– dialparties.agi: Extension 100 do not disturb is disabled
> dialparties.agi: extnum 100 has: cw: 1; hascfb: 0 [] hascfu: 0 []
– dialparties.agi: dbset CALLTRACE/100 to 103
– dialparties.agi: Filtered ARG3: 100
> dialparties.agi: NODEST: 600 adding M(auto-blkvm) to dialopts: TtrwWM(a uto-blkvm)
> dialparties.agi: NODEST: 600 blkvm enabled macro already in dialopts: T trwWM(auto-blkvm)
– <SIP/103-00000000>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/103-00000000”, “SIP/100,20,TtrwWM(au to-blkvm)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Called 100
== Extension Changed 100[ext-local] new state Ringing for Notify User 117
== Extension Changed 100[ext-local] new state Ringing for Notify User 110
== Extension Changed 100[ext-local] new state Ringing for Notify User 103
== Extension Changed 100[ext-local] new state Ringing for Notify User 101
– SIP/100-00000001 is ringing
– SIP/100-00000001 is ringing
– SIP/100-00000001 is ringing
– SIP/100-00000001 is ringing
– SIP/100-00000001 is ringing
– SIP/100-00000001 is ringing
– Nobody picked up in 20000 ms
== Extension Changed 100[ext-local] new state Idle for Notify User 117
== Extension Changed 100[ext-local] new state Idle for Notify User 110
== Extension Changed 100[ext-local] new state Idle for Notify User 103
== Extension Changed 100[ext-local] new state Idle for Notify User 101
– Executing [[email protected]:8] Set(“SIP/103-00000000”, “DIALSTATUS=NOANSWER”) in new stack
– Executing [[email protected]:9] GosubIf(“SIP/103-00000000”, “0?NOANSWER,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/103-00000000”, “RingGroupMethod=”) in new stack
– Executing [[email protected]:15] GotoIf(“SIP/103-00000000”, “0?nodest”) in ne w stack
– Executing [[email protected]:16] Set(“SIP/103-00000000”, “__NODEST=”) in new stack
– Executing [[email protected]:17] DBdel(“SIP/103-00000000”, “BLKVM/600/SIP/103 -00000000”) in new stack
– DBdel: family=BLKVM, key=600/SIP/103-00000000
– Executing [[email protected]:18] Goto(“SIP/103-00000000”, “ext-group,601,1”) in new stack
– Goto (ext-group,601,1)
– Executing [[email protected]:1] Macro(“SIP/103-00000000”, “user-callerid,”) i n new stack
– Executing [[email protected]:1] Set(“SIP/103-00000000”, “AMPUSER=103” ) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/103-00000000”, “0?report” ) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/103-00000000”, “0?Set(REA LCALLERIDNUM=103)”) in new stack
– Executing [[email protected]:4] Set(“SIP/103-00000000”, “AMPUSER=103” ) in new stack
– Executing [[email protected]:5] Set(“SIP/103-00000000”, “AMPUSERCIDNA ME=Anneline”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/103-00000000”, “0?report” ) in new stack
– Executing [[email protected]:7] Set(“SIP/103-00000000”, “AMPUSERCID=1 03”) in new stack
– Executing [[email protected]:8] Set(“SIP/103-00000000”, “CALLERID(all )=“Anneline” <103>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/103-00000000”, “0?Set(CHA NNEL(language)=)”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/103-00000000”, “0?contin ue”) in new stack
– Executing [[email protected]:11] Set(“SIP/103-00000000”, “__TTL=63”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/103-00000000”, “1?contin ue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/103-00000000”, “Using Call erID “xxxxxxx” <103>”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/103-00000000”, “0?skipdb”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/103-00000000”, “0?skipov”) in new stack
– Executing [[email protected]:4] Set(“SIP/103-00000000”, “__NODEST=”) in new s tack
– Executing [[email protected]:5] Set(“SIP/103-00000000”, “__BLKVM_OVERRIDE=BLK VM/601/SIP/103-00000000”) in new stack
– Executing [[email protected]:6] Set(“SIP/103-00000000”, “__BLKVM_BASE=601”) i n new stack
– Executing [[email protected]:7] Set(“SIP/103-00000000”, “DB(BLKVM/601/SIP/103 -00000000)=TRUE”) in new stack
– Executing [[email protected]:8] Set(“SIP/103-00000000”, “RRNODEST=”) in new s tack
– Executing [[email protected]:9] Set(“SIP/103-00000000”, “__NODEST=601”) in ne w stack
– Executing [[email protected]:10] Set(“SIP/103-00000000”, “RecordMethod=Group” ) in new stack
– Executing [[email protected]:11] Macro(“SIP/103-00000000”, “record-enable,101 -102-117-110,Group”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/103-00000000”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/103-00000000”, “recordingche ck,20140521-171154,1400685094.0”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
– <SIP/103-00000000>AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/103-00000000”, “”) in new stack
– Executing [[email protected]:12] Set(“SIP/103-00000000”, “RingGroupMethod=rin gall”) in new stack
– Executing [[email protected]:13] Macro(“SIP/103-00000000”, “dial,20,TtrwW,101 -102-117-110”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/103-00000000”, “1?dial”) in new st ack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/103-00000000”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is ‘xxxxxx’ number is '103’
dialparties.agi: Methodology of ring is ‘ringall’
– dialparties.agi: Added extension 101 to extension map
– dialparties.agi: Added extension 102 to extension map
– dialparties.agi: Added extension 117 to extension map
– dialparties.agi: Added extension 110 to extension map
– dialparties.agi: Extension 101 cf is disabled
– dialparties.agi: Extension 102 cf is disabled
– dialparties.agi: Extension 117 cf is disabled
– dialparties.agi: Extension 110 cf is disabled
– dialparties.agi: Extension 101 do not disturb is disabled
– dialparties.agi: Extension 102 do not disturb is disabled
– dialparties.agi: Extension 117 do not disturb is disabled
– dialparties.agi: Extension 110 do not disturb is disabled
> dialparties.agi: extnum 101 has: cw: 1; hascfb: 0 [] hascfu: 0 []
– dialparties.agi: dbset CALLTRACE/101 to 103
> dialparties.agi: extnum 102 has: cw: 1; hascfb: 0 [] hascfu: 0 []
– dialparties.agi: dbset CALLTRACE/102 to 103
> dialparties.agi: extnum 117 has: cw: 1; hascfb: 0 [] hascfu: 0 []
– dialparties.agi: dbset CALLTRACE/117 to 103
> dialparties.agi: extnum 110 has: cw: 1; hascfb: 0 [] hascfu: 0 []
– dialparties.agi: dbset CALLTRACE/110 to 103
– dialparties.agi: Filtered ARG3: 101-102-117-110
> dialparties.agi: NODEST: 601 adding M(auto-blkvm) to dialopts: TtrwWM(a uto-blkvm)
> dialparties.agi: NODEST: 601 blkvm enabled macro already in dialopts: T trwWM(auto-blkvm)
– <SIP/103-00000000>AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/103-00000000”, “SIP/101&SIP/102&SIP/ 117&SIP/110,20,TtrwWM(auto-blkvm)”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Called 101
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Called 117
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Called 110
== Extension Changed 101[ext-local] new state Ringing for Notify User 103
== Extension Changed 101[ext-local] new state Ringing for Notify User 100
== Extension Changed 101[ext-local] new state Ringing for Notify User 110
== Extension Changed 101[ext-local] new state Ringing for Notify User 117
== Extension Changed 117[ext-local] new state Ringing for Notify User 100
== Extension Changed 110[ext-local] new state Ringing for Notify User 103
== Extension Changed 110[ext-local] new state Ringing for Notify User 100
== Extension Changed 110[ext-local] new state Ringing for Notify User 101
== Extension Changed 110[ext-local] new state Ringing for Notify User 117
– SIP/110-00000004 is ringing
– SIP/101-00000002 is ringing
– SIP/117-00000003 is ringing
– SIP/110-00000004 is ringing
– SIP/101-00000002 is ringing
– SIP/117-00000003 is ringing
– SIP/110-00000004 is ringing
– SIP/101-00000002 is ringing
– SIP/117-00000003 is ringing
– SIP/110-00000004 is ringing
– SIP/101-00000002 is ringing
– SIP/117-00000003 is ringing
– SIP/110-00000004 is ringing
– SIP/101-00000002 is ringing
– SIP/117-00000003 is ringing
== Extension Changed 103[ext-local] new state Idle for Notify User 117
== Extension Changed 103[ext-local] new state Idle for Notify User 103
== Extension Changed 103[ext-local] new state Idle for Notify User 100
== Extension Changed 103[ext-local] new state Idle for Notify User 110
== Extension Changed 103[ext-local] new state Idle for Notify User 101
== Extension Changed 110[ext-local] new state Idle for Notify User 103
== Extension Changed 110[ext-local] new state Idle for Notify User 100
== Extension Changed 110[ext-local] new state Idle for Notify User 101
== Extension Changed 110[ext-local] new state Idle for Notify User 117
== Extension Changed 117[ext-local] new state Idle for Notify User 100
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/103-00000000’ in macro ‘dial’
== Extension Changed 101[ext-local] new state Idle for Notify User 103
== Extension Changed 101[ext-local] new state Idle for Notify User 100
== Extension Changed 101[ext-local] new state Idle for Notify User 110
== Extension Changed 101[ext-local] new state Idle for Notify User 117
== Spawn extension (ext-group, 601, 13) exited non-zero on ‘SIP/103-00000000’
– ast_get_srv: SRV lookup for ‘_sip._UDP.ISVOIP.NET’ mapped to host sip.isvoip.net, port xxxx
trixbox*CLI>

at the end it looks like its trying to map it to my sip trunk but no luck.

sorry again if this is vague - for asterix/trixbox im really a newby

I suggest you click on the support link above and let one of our engineers look at this for you.

You also need to get off of tribox, it is not supported, not for years.