Call don't hit to asterisk if get any busy signal from trunk

Asterisk Asterisk Support
1

Dear All,

I am running asterisk server of version 1.8. I am using eyebeam softphone registered to asterisk server as SIP extension. One gsm gateway is connected to asterisk as SIP trunk. One outbound route is created for this trunk.

call is from softphone to gsm gateway to gsm network.

Everything works fine other than the following problem…

When all ports in the gsm gw(or all-circuit are busy now-pls try later…message like this) is busy then call from softphone do not hit to asterisk for 100 seconds. But if I re-register softphone to asterisk then call hits to asterisk server. I think there should be any configuration options to avoid this.

Please help me out!

Regards,
Zahid Hasan
+8801732997739

2

There are no configuration options that should introduce such buggy behaviour. You need to provide enough diagnostic information to be able to work out what is really going wrong (certainly verbose console output and probably SIP traces). There might then be a way round.

I would advise trying with a non-soft SIP phone as I have had problems with X-Lite, the free version of the eyebeam.

Also you refer to trunks. This is a term used by Asterisk GUIs, rather than Asterisk. Are you using a GUI?

Are you sure you haven’t triggered some security software by making frequent failed calls? fail2ban is often used by Asterisk users, but this might be downstream.

3

Dear David55,
Thank you very much for your reply. You are right. I am using elastix 2.0.3 stable edition that contains freepbx 2.7.0.3 and asterisk 1.6.2.13. Anyway I have re-checked this and found that this only happens with h323 trunk(may be incorrectly configured). With my sip trunk this running very well. I think my h323 trunk configuratin is wrong. Actually I want to send call from my asterisk server(elastix 2.0.3) to quintum DX2030(in h323 mode)…Following is the sip debug message…

Asterisk server IP: 114.141.208.30
Quintum DX 2030 IP: 114.141.208.157 h323

SIP Debugging enabled

<— SIP read from UDP:114.141.208.29:12442 —>
INVITE sip:8801732997739@114.141.208.30 SIP/2.0
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-7a07951835209b67-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:11111101@114.141.208.29:12442
To: "8801732997739"sip:8801732997739@114.141.208.30
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 444

v=0
o=- 9 2 IN IP4 114.141.208.29
s=CounterPath eyeBeam 1.5
c=IN IP4 114.141.208.29
t=0 0
m=audio 36204 RTP/AVP 0 8 18 101
a=alt:1 4 : gNfQjagJ 0pZ2IUbd 114.141.208.29 36204
a=alt:2 3 : JV1/1uad wC7zTZ9d 172.16.125.189 36204
a=alt:3 2 : OHRlCjDX g3Ftkods 172.30.30.28 36204
a=alt:4 1 : JuGbPHtB rjGRmTio 192.168.56.133 36204
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (12 headers 15 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 114.141.208.29 : 12442 (no NAT)
Using INVITE request as basis request - OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
Found peer ‘11111101’ for ‘11111101’ from 114.141.208.29:12442

<— Reliably Transmitting (NAT) to 114.141.208.29:12442 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-7a07951835209b67-1—d8754z-;received=114.141.208.29;rport=12442
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
To: "8801732997739"sip:8801732997739@114.141.208.30;tag=as14740df1
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2fbbc3d0"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:114.141.208.29:12442 —>
ACK sip:8801732997739@114.141.208.30 SIP/2.0
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-7a07951835209b67-1—d8754z-;rport
To: "8801732997739"sip:8801732997739@114.141.208.30;tag=as14740df1
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 1 ACK
Content-Length: 0

------------->
— (7 headers 0 lines) —

<— SIP read from UDP:114.141.208.29:12442 —>
INVITE sip:8801732997739@114.141.208.30 SIP/2.0
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-8542d33f8f2b9a46-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:11111101@114.141.208.29:12442
To: "8801732997739"sip:8801732997739@114.141.208.30
From: “Tony"sip:11111101@114.141.208.30;tag=095a1d3b
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Authorization: Digest username=“11111101”,realm=“asterisk”,nonce=“2fbbc3d0”,uri="sip:8801732997739@114.141.208.30”,response=“bf6e5bd6b0b0dd0b4c032f8dbeaa1db6”,algorithm=MD5
Content-Length: 444

v=0
o=- 9 2 IN IP4 114.141.208.29
s=CounterPath eyeBeam 1.5
c=IN IP4 114.141.208.29
t=0 0
m=audio 36204 RTP/AVP 0 8 18 101
a=alt:1 4 : gNfQjagJ 0pZ2IUbd 114.141.208.29 36204
a=alt:2 3 : JV1/1uad wC7zTZ9d 172.16.125.189 36204
a=alt:3 2 : OHRlCjDX g3Ftkods 172.30.30.28 36204
a=alt:4 1 : JuGbPHtB rjGRmTio 192.168.56.133 36204
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
— (13 headers 15 lines) —
Sending to 114.141.208.29 : 12442 (NAT)
Using INVITE request as basis request - OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
Found peer ‘11111101’ for ‘11111101’ from 114.141.208.29:12442
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 114.141.208.29:36204
Looking for 8801732997739 in from-internal (domain 114.141.208.30)
list_route: hop: sip:11111101@114.141.208.29:12442

<— Transmitting (NAT) to 114.141.208.29:12442 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-8542d33f8f2b9a46-1—d8754z-;received=114.141.208.29;rport=12442
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
To: "8801732997739"sip:8801732997739@114.141.208.30
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:8801732997739@114.141.208.30
Content-Length: 0

<------------>
– Executing [8801732997739@from-internal:1] Macro(“SIP/11111101-0000000b”, “user-callerid,SKIPTTL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/11111101-0000000b”, “AMPUSER=11111101”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/11111101-0000000b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/11111101-0000000b”, “1?Set(REALCALLERIDNUM=11111101)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/11111101-0000000b”, “AMPUSER=11111101”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/11111101-0000000b”, “AMPUSERCIDNAME=Tony”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/11111101-0000000b”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/11111101-0000000b”, “AMPUSERCID=11111101”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/11111101-0000000b”, “CALLERID(all)=“Tony” <11111101>”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“SIP/11111101-0000000b”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/11111101-0000000b”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/11111101-0000000b”, “Using CallerID “Tony” <11111101>”) in new stack
– Executing [8801732997739@from-internal:2] ExecIf(“SIP/11111101-0000000b”, “1?Set(TRUNKCIDOVERRIDE=33333333)”) in new stack
– Executing [8801732997739@from-internal:3] Set(“SIP/11111101-0000000b”, “_NODEST=”) in new stack
– Executing [8801732997739@from-internal:4] Macro(“SIP/11111101-0000000b”, “record-enable,11111101,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/11111101-0000000b”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] ExecIf(“SIP/11111101-0000000b”, “0?MacroExit()”) in new stack
– Executing [s@macro-record-enable:5] GotoIf(“SIP/11111101-0000000b”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [s@macro-record-enable:15] GotoIf(“SIP/11111101-0000000b”, “0?IN”) in new stack
– Executing [s@macro-record-enable:16] ExecIf(“SIP/11111101-0000000b”, “1?MacroExit()”) in new stack
– Executing [8801732997739@from-internal:5] Macro(“SIP/11111101-0000000b”, “dialout-trunk,3,8801732997739,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/11111101-0000000b”, “DIAL_TRUNK=3”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/11111101-0000000b”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/11111101-0000000b”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/11111101-0000000b”, “DIAL_NUMBER=8801732997739”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/11111101-0000000b”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/11111101-0000000b”, “OUTBOUND_GROUP=OUT_3”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/11111101-0000000b”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/11111101-0000000b”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/11111101-0000000b”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/11111101-0000000b”, “outbound-callerid,3”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/11111101-0000000b”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/11111101-0000000b”, “0?Set(REALCALLERIDNUM=11111101)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/11111101-0000000b”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/11111101-0000000b”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/11111101-0000000b”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/11111101-0000000b”, “TRUNKOUTCID=33333333”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/11111101-0000000b”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/11111101-0000000b”, “1?Set(CALLERID(all)=33333333)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/11111101-0000000b”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/11111101-0000000b”, “1?Set(CALLERID(all)=33333333)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/11111101-0000000b”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/11111101-0000000b”, “1?AGI(fixlocalprefix)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== fixlocalprefix: Dialpattern . matched. 8801732997739 -> 8801732997739
– <SIP/11111101-0000000b>AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/11111101-0000000b”, “OUTNUM=8801732997739”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/11111101-0000000b”, “custom=AMP”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/11111101-0000000b”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/11111101-0000000b”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/11111101-0000000b”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/11111101-0000000b”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/11111101-0000000b”, “1?customtrunk”) in new stack
– Goto (macro-dialout-trunk,s,22)
– Executing [s@macro-dialout-trunk:22] Set(“SIP/11111101-0000000b”, “pre_num=AMP:H323/”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/11111101-0000000b”, “the_num=OUTNUM”) in new stack
– Executing [s@macro-dialout-trunk:24] Set(“SIP/11111101-0000000b”, "post_num=@114.141.208.157") in new stack
– Executing [s@macro-dialout-trunk:25] GotoIf(“SIP/11111101-0000000b”, “1?outnum:skipoutnum”) in new stack
– Goto (macro-dialout-trunk,s,26)
– Executing [s@macro-dialout-trunk:26] Set(“SIP/11111101-0000000b”, “the_num=8801732997739”) in new stack
– Executing [s@macro-dialout-trunk:27] Dial(“SIP/11111101-0000000b”, “H323/8801732997739@114.141.208.157,300,”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:28] NoOp(“SIP/11111101-0000000b”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 66”) in new stack
– Executing [s@macro-dialout-trunk:29] Goto(“SIP/11111101-0000000b”, “s-CHANUNAVAIL,1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/11111101-0000000b”, “RC=66”) in new stack
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/11111101-0000000b”, “66,1”) in new stack
– Goto (macro-dialout-trunk,66,1)
– Executing [66@macro-dialout-trunk:1] Goto(“SIP/11111101-0000000b”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/11111101-0000000b”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/11111101-0000000b”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/11111101-0000000b”, “CALLERID(number)=11111101”) in new stack
– Executing [8801732997739@from-internal:6] Macro(“SIP/11111101-0000000b”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/11111101-0000000b”, “”) in new stack
Audio is at 114.141.208.30 port 10644
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 114.141.208.29:12442 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-8542d33f8f2b9a46-1—d8754z-;received=114.141.208.29;rport=12442
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
To: "8801732997739"sip:8801732997739@114.141.208.30;tag=as68393c63
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:8801732997739@114.141.208.30
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 207365778 207365778 IN IP4 114.141.208.30
s=Asterisk PBX 1.6.2.13
c=IN IP4 114.141.208.30
t=0 0
m=audio 10644 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/11111101-0000000b”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/11111101-0000000b”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/11111101-0000000b”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/11111101-0000000b> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
– <SIP/11111101-0000000b> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
– Executing [s@macro-outisbusy:5] Congestion(“SIP/11111101-0000000b”, “20”) in new stack

<— Reliably Transmitting (NAT) to 114.141.208.29:12442 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-8542d33f8f2b9a46-1—d8754z-;received=114.141.208.29;rport=12442
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
To: "8801732997739"sip:8801732997739@114.141.208.30;tag=as68393c63
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-Asterisk-HangupCause: Channel not implemented
X-Asterisk-HangupCauseCode: 66
Content-Length: 0

<------------>
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on ‘SIP/11111101-0000000b’ in macro ‘outisbusy’
== Spawn extension (from-internal, 8801732997739, 6) exited non-zero on ‘SIP/11111101-0000000b’
– Executing [h@from-internal:1] Macro(“SIP/11111101-0000000b”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/11111101-0000000b”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] NoOp(“SIP/11111101-0000000b”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/11111101-0000000b”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/11111101-0000000b”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,10)
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/11111101-0000000b”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,12)
– Executing [s@macro-hangupcall:12] Hangup(“SIP/11111101-0000000b”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/11111101-0000000b’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/11111101-0000000b’

<— SIP read from UDP:114.141.208.29:12442 —>
ACK sip:8801732997739@114.141.208.30 SIP/2.0
Via: SIP/2.0/UDP 114.141.208.29:12442;branch=z9hG4bK-d8754z-8542d33f8f2b9a46-1—d8754z-;rport
To: "8801732997739"sip:8801732997739@114.141.208.30;tag=as68393c63
From: "Tony"sip:11111101@114.141.208.30;tag=095a1d3b
Call-ID: OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.
CSeq: 2 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘OWQyZDRlZTVmZjhmMjU2YTVhZjVmNjg2MzdjOGE2MWE.’ Method: ACK
switchCLI> sip debug off
No such command ‘sip debug off’ (type ‘core show help sip debug off’ for other possible commands)
switchCLI> exit

I am looking forward to hearing from you.

Regards,
Zahid

4

You are not getting busy, you are getting channel unavailable, with a more specific diagnostic of :
Cause No. 66 - channel type not implemented.
This cause indicates that the equipment sending this cause does not support the channel type requested.

Do you actually have the H.323 module loaded?

5

Hello David55,

There is a default h323.conf file in /etc/asterisk/ directory. But when I enter the command “core show channeltypes” no h323 module found in the output. Then I copied ooh323.conf.sample to ooh323.conf. Now I show the following output from “core show channeltypes”:
switch*CLI> core show channeltypes
Type Description Devicestate Indications Transfer

Bridge Bridge Interaction Channel no no no
Agent Call Agent Proxy Channel yes yes no
MGCP Media Gateway Control Protocol (MGCP) yes yes no
SIP Session Initiation Protocol (SIP) yes yes yes
Phone Standard Linux Telephony API Driver no yes no
OOH323 Objective Systems H323 Channel Driver no yes no
DAHDI DAHDI Telephony Driver w/PRI & MFC/R2 no yes no
WOOMERA Woomera Channel Driver no yes yes
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes
USTM UNISTIM Channel Driver no yes no
Local Local Proxy Channel Driver yes yes no

11 channel drivers registered.

And following is the configuration in ooh323.conf

[root@switch asterisk]# more ooh323.conf
; ------------------------------------------------------------------------------

; — ******* IMPORTANT NOTE ***********
; —
; — This module is currently unsupported. Use it at your own risk.
; —
; ------------------------------------------------------------------------------

; Objective System’s H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as “dynamic” is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H
323
; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/exten@ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.

[general]
;Define the asetrisk server h323 endpoint

;The port asterisk should listen for incoming H323 connections.
;Default - 1720
port=1720

;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it’s own
bindaddr=114.141.208.30

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
faststart=yes
h245tunneling=yes

;Whether media wait for connect
;Default - No
;mediawaitforconnect=yes

;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
e164=100

;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE

;Location for H323 log file
;Default - /var/log/asterisk/h323_log
logfile=/var/log/asterisk/h323_log

;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition

;Sets default context all clients will be placed in.
;Default - default
context=default

;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we’re not on hold

;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay

;amaflags = default

;The account code used by default for all clients.
;accountcode=h3230101

;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
;allow=gsm
allow=g729

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833

; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout

;

;Define users here
;Section header is extension
[myuser1]
type=user
context=context1
disallow=all
allow=gsm
allow=ulaw

[mypeer1]
type=peer
context=context2
ip=a.b.c.d ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=101

[myfriend1]
type=friend
context=default
ip=10.0.0.82 ; UPDATE with appropriate ip address
port=1820 ; UPDATE with appropriate port
disallow=all
allow=ulaw
e164=12345
rtptimeout=60
dtmfmode=rfc2833

I ALSO HAVE CERATED A CUSTOM TRUNK NAMED “DX2030” FROM ELASTIX GUI LIKE "OH323/$OUTNUM$@114.141.208.157"
AND CREATED AN OUTBOUND ROUTE FOR THIS TRUNK. BUT CALL DO NOT HIT TO QUINTUM.

PLEASE HELP ME OUT!

Regards,
Zahid