Simple SIP-TRUNK question

Asterisk Asterisk Support
1

i searched and tried many dial plans, but all fails ,so i made this noob topic

i created a sip trunk to my VOIP operator
and sip trunk is registered ( checked this with Wireshark)
all extensions can make internal calls ,and everything is working fine

when i try to make an outbound calls (using my sip trunk) i always see a cancel sip message from the VOIP operator

dial plans i used:
9.
9|08111XXXXXX
9|050XXXXXXX

my sip number is : 009668111XXXXXX
local operator numbers are: 0096650XXXXXXX (mobile ) or 009661XXXXXXX (PSTN) ( or dial them with 0 instead of 00966)

if i try to call from my mobile to the SIP account (incoming call to asterisk) i receive :
SIP/SDP Request: INVITE sip:s@77.30.62.74:5060, with session description
its showing “s” instead of my " mobile number 050xxxxxxx

can anyone point me in the right direction???
hope someone can help :smile:

2

CANCEL can only be sent by the client. The service provider is server in this context. It can’t be sending CANCEL.

The actual dialogue would help.

3

thanks david for the reply

can you please let me know if my dial plans are good?
why 9. is failing to generate calls
i will try another Client ,and post back

4

You’ve only included phone dial plans, not Asterisk ones, and you haven’t said what type of phone this is.

On the incoming call, you don’t have a DID trunk, even if some people call all PSTN gateway trunks DID ones.

5

[quote=“david55”]You’ve only included phone dial plans, not Asterisk ones, and you haven’t said what type of phone this is.

On the incoming call, you don’t have a DID trunk, even if some people call all PSTN gateway trunks DID ones.[/quote]

Hi David
i am using X-lite Softphone
i did some tests again , and i noticed that when i try to make call >> i am getting a " all circuits are busy now ,please try again later" , and then it gives the BUSY TONE , and i will have 1 option= hangup ( which is the CANCEL message i am seeing on WireShark )

on the incoming call >> i didn’t specify any DID , i just forwarded all calls to my 5065 extension

i didn’t configure anything on the Dialplans in the Extensions.conf file , as i said i am following the book “trixbox2_without_tears” to configure this ,and i didn’t see anything related to extensions.conf changes

what am i missing?
Help anyone?

6

Guys come on , its a simple Question , i bet all of you can answer it , you already have some system Running witha registered SIP TRUNK

can anyone give me a hint or something , google isn’t helping me ,adn i tried many configs & number all same failure

please HELP!!!

7

The s on the incoming call is because of the type of service you have bought from your provider, and, in any case, that would be the DID number, not the caller ID.

The best way of setting up the trunk side is to ask your service provider. If they are providing a PABX service they should have a standard Asterisk configuration.

Without your Asterisk dialplan, and sip.conf contents, and verbose asterisk CLI output, it really is difficult to know what you are doing wrong.

8

[quote=“david55”]The s on the incoming call is because of the type of service you have bought from your provider, and, in any case, that would be the DID number, not the caller ID.

The best way of setting up the trunk side is to ask your service provider. If they are providing a PABX service they should have a standard Asterisk configuration.

Without your Asterisk dialplan, and sip.conf contents, and verbose asterisk CLI output, it really is difficult to know what you are doing wrong.[/quote]
here is a outgoing call CLI with verbose 10

asterisk*CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Executing [90500432981@from-internal:1] Macro("SIP/400000-0000001a", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/400000-0000001a", "AMPUSER=400000") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/400000-0000001a", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/400000-0000001a", "1?Set(REALCALLERIDNUM=400000)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/400000-0000001a", "AMPUSER=400000") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/400000-0000001a", "AMPUSERCIDNAME=400000") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/400000-0000001a", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/400000-0000001a", "AMPUSERCID=400000") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/400000-0000001a", "CALLERID(all)="400000" <400000>") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/400000-0000001a", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/400000-0000001a", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("SIP/400000-0000001a", "Using CallerID "400000" <400000>") in new stack -- Executing [90500432981@from-internal:2] Set("SIP/400000-0000001a", "_NODEST=") in new stack -- Executing [90500432981@from-internal:3] Macro("SIP/400000-0000001a", "record-enable,400000,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/400000-0000001a", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/400000-0000001a", "recordingcheck,20110520-022331,1305847411.26") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck,20110520-022331,1305847411.26: Outbound recording not enabled -- <SIP/400000-0000001a>AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:5] MacroExit("SIP/400000-0000001a", "") in new stack -- Executing [90500432981@from-internal:4] Macro("SIP/400000-0000001a", "dialout-trunk,2,0500432981,,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/400000-0000001a", "DIAL_TRUNK=2") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/400000-0000001a", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/400000-0000001a", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/400000-0000001a", "DIAL_NUMBER=0500432981") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/400000-0000001a", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/400000-0000001a", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/400000-0000001a", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/400000-0000001a", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/400000-0000001a", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/400000-0000001a", "outbound-callerid,2") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/400000-0000001a", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/400000-0000001a", "0?Set(REALCALLERIDNUM=400000)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/400000-0000001a", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/400000-0000001a", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/400000-0000001a", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/400000-0000001a", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/400000-0000001a", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/400000-0000001a", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/400000-0000001a", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/400000-0000001a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/400000-0000001a", "1?AGI(fixlocalprefix)") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Using pattern 9|2020 > fixlocalprefix: Using pattern 9|XXXXXXXXXX > fixlocalprefix: Using pattern 9|XXXXXXXXXXX -- <SIP/400000-0000001a>AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:13] Set("SIP/400000-0000001a", "OUTNUM=0500432981") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/400000-0000001a", "custom=SIP/sipXTrunk") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/400000-0000001a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/400000-0000001a", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/400000-0000001a", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/400000-0000001a", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/400000-0000001a", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/400000-0000001a", "SIP/sipXTrunk/0500432981,300,") 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 sipXTrunk/0500432981 << [ TYPE: Control (4) SUBCLASS: Congestion (8) ] [SIP/sipXTrunk-0000001b] -- SIP/sipXTrunk-0000001b is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] Goto("SIP/400000-0000001a", "s-CONGESTION,1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/400000-0000001a", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,3) -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/400000-0000001a", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing [90500432981@from-internal:5] Macro("SIP/400000-0000001a", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Playback("SIP/400000-0000001a", "all-circuits-busy-now,noanswer") in new stack -- <SIP/400000-0000001a> Playing 'all-circuits-busy-now.ulaw' (language 'en') << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/400000-0000001a] -- Executing [s@macro-outisbusy:2] Playback("SIP/400000-0000001a", "pls-try-call-later,noanswer") in new stack -- <SIP/400000-0000001a> Playing 'pls-try-call-later.ulaw' (language 'en') << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/400000-0000001a] -- Executing [s@macro-outisbusy:3] Macro("SIP/400000-0000001a", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/400000-0000001a", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/400000-0000001a", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/400000-0000001a", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/400000-0000001a", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/400000-0000001a' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/400000-0000001a' in macro 'outisbusy' == Spawn extension (from-internal, 90500432981, 5) exited non-zero on 'SIP/400000-0000001a' -- Executing [h@from-internal:1] Macro("SIP/400000-0000001a", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/400000-0000001a", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/400000-0000001a", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/400000-0000001a", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/400000-0000001a", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/400000-0000001a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/400000-0000001a' asterisk*CLI>

9

That is a complex custom dialplan with custom AGI scripting. You need to ask the person who wrote it, or, if it is a standard GUI, and there is no viable support forum for that GUI, tell us which GUI it is.

10

this is the default one that came in my FreePBX asterisk

i didnt change any thing in it

11

In that case, you have a question about the FreePBX GUI, not about Asterisk.

Some people here have FreePBX experience, but as AsteriskNow is currently based on FreePBX, you may find that there are more on the AsteriskNow forum.