[HELP] SIP ext. to IAX ext. can not make a call


#1

I have latest ver. A@H, set it up from ISO and made couple ext.thru FreePBX GUI. Some are SIP, some are IAX. were is no problem calling SIP to SIP or IAX to IAX, but I can not make call SIP to IAX or IAX to SIP. Why? Am I missing some setting or something?
The only things I have done here after install, enabled all modules under Module Admin, created couple ext (sip 7925, 2000, 1000 and IAX2 4000, 4001 and so on) and enabled couple codecs in sip.conf and iax.conf and nothing else. I am using ATA adapter (SIP Grandstream 496), BOL SIPPhone (softphone) and Idefisk (IAX softphone).
Please help me…

Thanks


#2

This is CLI call IAX to SIP (ATA rings, but no connection when pickup handset)

– Accepting AUTHENTICATED call from 192.168.1.100:

requested format = gsm,
requested prefs = (),
actual format = speex,
host prefs = (speex|alaw|ilbc|gsm),
priority = mine
– Executing Macro(“IAX2/4001-1”, “exten-vm|2000|2000”) in new stack
– Executing Macro(“IAX2/4001-1”, “user-callerid”) in new stack
– Executing Set(“IAX2/4001-1”, “AMPUSER=4001”) in new stack
– Executing Set(“IAX2/4001-1”, “EMERGENCYCID=”) in new stack
– Executing Set(“IAX2/4001-1”, “AMPUSERCIDNAME=IAXLaptop”) in new stack
– Executing GotoIf(“IAX2/4001-1”, “0?6”) in new stack
– Executing Set(“IAX2/4001-1”, “CALLERID(all)=“IAXLaptop” <4001>”) in new stack
– Executing NoOp(“IAX2/4001-1”, “Using CallerID “IAXLaptop” <4001>”) in new stack
– Executing Set(“IAX2/4001-1”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Macro(“IAX2/4001-1”, “record-enable|2000|IN”) in new stack
– Executing GotoIf(“IAX2/4001-1”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“IAX2/4001-1”, “recordingcheck|20060425-124231|1145986951.4”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060425-124231|1145986951.4: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“IAX2/4001-1”, “No recording needed”) in new stack
– Executing Macro(“IAX2/4001-1”, “dial|15|tr|2000”) in new stack
– Executing AGI(“IAX2/4001-1”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
– dialparties.agi: priority = 1
– dialparties.agi: callingani2 = 0
– dialparties.agi: accountcode =
– dialparties.agi: channel = IAX2/4001-1
– dialparties.agi: callerid = 4001
– dialparties.agi: context = macro-dial
– dialparties.agi: callington = 0
– dialparties.agi: dnid = 2000
– dialparties.agi: request = dialparties.agi
– dialparties.agi: calleridname = IAXLaptop
– dialparties.agi: extension = s
– dialparties.agi: language = en
– dialparties.agi: uniqueid = 1145986951.4
– dialparties.agi: callingpres = 1
– dialparties.agi: type = IAX2
– dialparties.agi: rdnis = unknown
– dialparties.agi: callingtns = 0
– dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name is ‘IAXLaptop’ number is '4001’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 2000 to extension map
– dialparties.agi: Extension 2000 cf is disabled
– dialparties.agi: Extension 2000 do not disturb is disabled
– dialparties.agi: Checking CW and CFB status for extension 2000
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
– dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager ‘admin’ logged off from 127.0.0.1
dialparties.agi: Extension 2000 is available…skipping checks
– dialparties.agi: DbSet CALLTRACE/2000 to 4001
– AGI Script dialparties.agi completed, returning 0
– Executing Dial(“IAX2/4001-1”, “SIP/2000|15|tr”) in new stack
– Called 2000
– SIP/2000-0af0 is ringing

Try to answer the call and this happens:

– SIP/2000-0af0 answered IAX2/4001-1
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘IAX2/4001-1’ in macro ‘dial’
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘IAX2/4001-1’ in macro ‘exten-vm’
== Spawn extension (macro-dial, s, 10) exited non-zero on ‘IAX2/4001-1’
– Hungup 'IAX2/4001-1’
asterisk1*CLI>


#3

This is SIP(ATA Grandsteram) to IAX (Idefisk)

– Executing Macro(“SIP/2000-2382”, “exten-vm|4001|4001”) in new stack
– Executing Macro(“SIP/2000-2382”, “user-callerid”) in new stack
– Executing Set(“SIP/2000-2382”, “AMPUSER=2000”) in new stack
– Executing Set(“SIP/2000-2382”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/2000-2382”, “AMPUSERCIDNAME=SIP2000”) in new stack
– Executing GotoIf(“SIP/2000-2382”, “0?6”) in new stack
– Executing Set(“SIP/2000-2382”, “CALLERID(all)=“SIP2000” <2000>”) in new stack
– Executing NoOp(“SIP/2000-2382”, “Using CallerID “SIP2000” <2000>”) in new stack
– Executing Set(“SIP/2000-2382”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Macro(“SIP/2000-2382”, “record-enable|4001|IN”) in new stack
– Executing GotoIf(“SIP/2000-2382”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“SIP/2000-2382”, “recordingcheck|20060425-123607|1145986567.0”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060425-123607|1145986567.0: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/2000-2382”, “No recording needed”) in new stack
– Executing Macro(“SIP/2000-2382”, “dial|15|tr|4001”) in new stack
– Executing AGI(“SIP/2000-2382”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
– dialparties.agi: priority = 1
– dialparties.agi: callingani2 = 0
– dialparties.agi: accountcode =
– dialparties.agi: channel = SIP/2000-2382
– dialparties.agi: callerid = 2000
– dialparties.agi: context = macro-dial
– dialparties.agi: callington = 0
– dialparties.agi: dnid = 4001
– dialparties.agi: request = dialparties.agi
– dialparties.agi: calleridname = SIP2000
– dialparties.agi: extension = s
– dialparties.agi: language = en
– dialparties.agi: uniqueid = 1145986567.0
– dialparties.agi: callingpres = 0
– dialparties.agi: type = SIP
– dialparties.agi: rdnis = unknown
– dialparties.agi: callingtns = 0
– dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name is ‘SIP2000’ number is '2000’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 4001 to extension map
– dialparties.agi: Extension 4001 cf is disabled
– dialparties.agi: Extension 4001 do not disturb is disabled
– dialparties.agi: Checking CW and CFB status for extension 4001
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
– dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager ‘admin’ logged off from 127.0.0.1
dialparties.agi: Extension 4001 is available…skipping checks
– dialparties.agi: DbSet CALLTRACE/4001 to 2000
– AGI Script dialparties.agi completed, returning 0
– Executing Dial(“SIP/2000-2382”, “IAX2/4001|15|tr”) in new stack
– Hungup ‘IAX2/4001-2’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing GotoIf(“SIP/2000-2382”, “0?s-CHANUNAVAIL|1”) in new stack
– Executing GotoIf(“SIP/2000-2382”, “0?s-CHANUNAVAIL|1”) in new stack
– Executing NoOp(“SIP/2000-2382”, “Sending to Voicemail box 4001”) in new stack
– Executing Macro(“SIP/2000-2382”, “vm|4001|CHANUNAVAIL”) in new stack
– Executing Macro(“SIP/2000-2382”, “user-callerid”) in new stack
– Executing Set(“SIP/2000-2382”, “AMPUSER=2000”) in new stack
– Executing Set(“SIP/2000-2382”, “EMERGENCYCID=”) in new stack
– Executing Set(“SIP/2000-2382”, “AMPUSERCIDNAME=SIP2000”) in new stack
– Executing GotoIf(“SIP/2000-2382”, “0?6”) in new stack
– Executing Set(“SIP/2000-2382”, “CALLERID(all)=“SIP2000” <2000>”) in new stack
– Executing NoOp(“SIP/2000-2382”, “Using CallerID “SIP2000” <2000>”) in new stack
– Executing Goto(“SIP/2000-2382”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-vm,s-CHANUNAVAIL,1)
– Executing Macro(“SIP/2000-2382”, “get-vmcontext|4001”) in new stack
– Executing Set(“SIP/2000-2382”, “VMCONTEXT=default”) in new stack
– Executing GotoIf(“SIP/2000-2382”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing NoOp(“SIP/2000-2382”, “”) in new stack
– Executing VoiceMail(“SIP/2000-2382”, “u4001@default”:wink: in new stack
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2000-2382’ in macro ‘vm’
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2000-2382’ in macro ‘exten-vm’
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2000-2382’
asterisk1*CLI>

Busy signal on ATA, no ringing on IAX softphone


#4

Post that again with sip debug and iax2 debug.

TheLostPacket


#5

sorry, I am pretty new to this. What do you meen SIP and IAX2 debug? configs?


#6

At the asterisk cli> prompt

sip debug
iax2 debug
set verbose 5

Then send a test call and log the output.

TheLostPacket


#7

Connected to Asterisk 1.2.7.1 currently running on asterisk1 (pid = 2579)
asterisk1*CLI>
Verbosity is at least 5

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI> sip debug

asterisk1*CLI>
SIP Debugging re-enabled

e[Kasterisk1*CLI> iax
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 13276 DCall: 00000 [192.168.1.103:4569]
USERNAME : 4000
REFRESH : 60

Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00007ms SCall: 00004 DCall: 13276 [192.168.1.103:4569]
AUTHMETHODS : 3
CHALLENGE : 122114588
USERNAME : 4000

e[Kasterisk1*CLI> iax
Rx-Frame Retry[ No] – OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 13276 DCall: 00004 [192.168.1.103:4569]
USERNAME : 4000
REFRESH : 60
MD5 RESULT : 8f8a6ed6dc26da2755cf3bd000b67330

Tx-Frame Retry[000] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK
Timestamp: 00013ms SCall: 00004 DCall: 13276 [192.168.1.103:4569]
USERNAME : 4000
DATE TIME : 2006-04-26 13:35:32
REFRESH : 60
APPARENT ADDRES : IPV4 192.168.1.103:4569
MESSAGE COUNT : 0
CALLING NUMBER : 4000
CALLING NAME : IAX

e[Kasterisk1*CLI> iax
Rx-Frame Retry[ No] – OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00013ms SCall: 13276 DCall: 00004 [192.168.1.103:4569]

e[Kasterisk1*CLI> iax2deb

<-- SIP read from 192.168.1.102:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI> iax2debuge[Ke[Ke[Ke[Ke[K debug

asterisk1*CLI>
IAX2 Debugging Enabled

e[Kasterisk1*CLI> set verbose 5

asterisk1*CLI>
Verbosity is at least 5

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:
INVITE sip:4001@192.168.1.101;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca36000012bdffff

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone

Contact: sip:2000@192.168.1.102;user=phone

Supported: replaces

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18713 INVITE

User-Agent: Grandstream HT496 1.0.2.16

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

C
e[Kasterisk1*CLI>
ontent-Type: application/sdp

Content-Length: 259

v=0

o=2000 8000 8000 IN IP4 192.168.1.102

s=SIP Call

c=IN IP4 192.168.1.102

t=0 0

m=audio 5004 RTP/AVP 4 97 101

a=sendrecv

a=rtpmap:4 G723/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=20

a=ptime:600

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-11

— (13 headers 13 lines)—
Using INVITE request as basis request - 1e29ffff5b2bffff@192.168.1.102
Sending to 192.168.1.102 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.102:5060:
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca36000012bdffff;received=192.168.1.102

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone;tag=as4a900d09

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18713 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: sip:4001@192.168.1.101

Proxy-Authenticate: Digest realm=“asterisk”, nonce=“23618af7”

Content-Length: 0


Scheduling destruction of call ‘1e29ffff5b2bffff@192.168.1.102’ in 15000 ms
Found user ‘2000’

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:
ACK sip:4001@192.168.1.101;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKca36000012bdffff

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone;tag=as4a900d09

Contact: sip:2000@192.168.1.102;user=phone

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18713 ACK

User-Agent: Grandstream HT496 1.0.2.16

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Content-Length: 0

— (11 headers 0 lines)—

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:
INVITE sip:4001@192.168.1.101;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK6e920000fea9ffff

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone

Contact: sip:2000@192.168.1.102;user=phone

Supported: replaces

Proxy-Authorization: Digest username=“2000”, realm=“asterisk”, algorithm=MD5, uri="sip:4001@192.168.1.101;user=phone", nonce=“23618af7”, response=“1781873eac58b5c3d209433abe6803e9”

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18714 INVITE

User-Agent: Grandstream HT496 1.0.2.16

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Content-Type: application/sdp

Content-Length: 259

v=0

o=2000 8000 8001 IN IP4 192.168.1.102

s=SIP Call

c=IN IP4 192.168.1.102

t=0 0

m=audio 5004 RTP/AVP 4 97 101

a=sendrecv

a=rtpmap:4 G723/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=20

a=ptime:600

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-11

— (14 headers 13 lines)—
Using INVITE request as basis request - 1e29ffff5b2bffff@192.168.1.102
Sending to 192.168.1.102 : 5060 (non-NAT)
Found user '2000’
Found RTP audio format 4
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.102:5004
Found description format G723
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x40d (g723|ulaw|alaw|ilbc), peer - audio=0x401 (g723|ilbc)/video=0x0 (nothing), combined - 0x401 (g723|ilbc)

e[Kasterisk1*CLI>
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4001 in from-internal (domain 192.168.1.101)
list_route: hop: sip:2000@192.168.1.102;user=phone
Transmitting (no NAT) to 192.168.1.102:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK6e920000fea9ffff;received=192.168.1.102

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18714 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: sip:4001@192.168.1.101

Content-Length: 0


e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mexten-vm|4001|4001e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40muser-calleride[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mAMPUSER=2000e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mEMERGENCYCID=e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mAMPUSERCIDNAME=SIP2000e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?6e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mCALLERID(all)=“SIP2000” <2000>e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mUsing CallerID “SIP2000” <2000>e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mFROMCONTEXT=exten-vme[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mrecord-enable|4001|INe[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0 > 0?2:4e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Goto (macro-record-enable,s,4)

e[Kasterisk1*CLI>
– Executing e[1;36;40mAGIe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mrecordingcheck|20060426-133609|1146076569.31e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

e[Kasterisk1*CLI>
recordingcheck|20060426-133609|1146076569.31: Inbound recording not enabled

e[Kasterisk1*CLI>
– AGI Script recordingcheck completed, returning 0
– Executing e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mNo recording needede[0;37;40m”) in new stack
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mdial|15|tr|4001e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mAGIe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mdialparties.agie[0;37;40m”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

e[Kasterisk1*CLI>
– dialparties.agi: priority = 1

e[Kasterisk1*CLI>
– dialparties.agi: callingani2 = 0

e[Kasterisk1*CLI>
– dialparties.agi: accountcode =

e[Kasterisk1*CLI>
– dialparties.agi: channel = SIP/2000-80e4

e[Kasterisk1*CLI>
– dialparties.agi: callerid = 2000

e[Kasterisk1*CLI>
– dialparties.agi: context = macro-dial

e[Kasterisk1*CLI>
– dialparties.agi: callington = 0

e[Kasterisk1*CLI>
– dialparties.agi: dnid = 4001

e[Kasterisk1*CLI>
– dialparties.agi: request = dialparties.agi

e[Kasterisk1*CLI>
– dialparties.agi: calleridname = SIP2000

e[Kasterisk1*CLI>
– dialparties.agi: extension = s

e[Kasterisk1*CLI>
– dialparties.agi: language = en

e[Kasterisk1*CLI>
– dialparties.agi: uniqueid = 1146076569.31

e[Kasterisk1*CLI>
– dialparties.agi: callingpres = 0

e[Kasterisk1*CLI>
– dialparties.agi: type = SIP

e[Kasterisk1*CLI>
– dialparties.agi: rdnis = unknown

e[Kasterisk1*CLI>
– dialparties.agi: callingtns = 0

e[Kasterisk1*CLI>
– dialparties.agi: enhanced = 0.0

e[Kasterisk1*CLI>
dialparties.agi: Caller ID name is ‘SIP2000’ number is ‘2000’

e[Kasterisk1*CLI>
dialparties.agi: Methodology of ring is ‘none’

e[Kasterisk1*CLI>
– dialparties.agi: Added extension 4001 to extension map

e[Kasterisk1*CLI>
– dialparties.agi: Extension 4001 cf is disabled

e[Kasterisk1*CLI>
– dialparties.agi: Extension 4001 do not disturb is disabled

e[Kasterisk1*CLI>
> dialparties.agi: extnum: 4001

e[Kasterisk1*CLI>
> dialparties.agi: exthascw: 0

e[Kasterisk1*CLI>
> dialparties.agi: exthascfb: 0

e[Kasterisk1*CLI>
> dialparties.agi: extcfb:

e[Kasterisk1*CLI>
– dialparties.agi: Checking CW and CFB status for extension 4001

e[Kasterisk1*CLI>

== Parsing ‘/etc/asterisk/manager.conf’: Found

e[Kasterisk1*CLI>

== Parsing ‘/etc/asterisk/manager_custom.conf’: Found

e[Kasterisk1*CLI>
== Manager ‘admin’ logged on from 127.0.0.1

e[Kasterisk1*CLI>
– dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS

e[Kasterisk1*CLI>
== Manager ‘admin’ logged off from 127.0.0.1

e[Kasterisk1*CLI>
> dialparties.agi: extstate: 0

e[Kasterisk1*CLI>
dialparties.agi: Extension 4001 is available…skipping checks

e[Kasterisk1*CLI>
– dialparties.agi: DbSet CALLTRACE/4001 to 2000

e[Kasterisk1*CLI>
– AGI Script dialparties.agi completed, returning 0
– Executing e[1;36;40mDiale[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mIAX2/4001|15|tre[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Hungup ‘IAX2/4001-2’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?s-CHANUNAVAIL|1e[0;37;40m”) in new stack
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?s-CHANUNAVAIL|1e[0;37;40m”) in new stack
– Executing e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mSending to Voicemail box 4001e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: HANGUP
Timestamp: 00005ms SCall: 00002 DCall: 00000 [192.168.1.100:4569]
CAUSE CODE : 0

e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mvm|4001|CHANUNAVAILe[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40muser-calleride[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mAMPUSER=2000e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mEMERGENCYCID=e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mAMPUSERCIDNAME=SIP2000e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?6e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mCALLERID(all)=“SIP2000” <2000>e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mUsing CallerID “SIP2000” <2000>e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00005ms SCall: 00834 DCall: 00002 [192.168.1.100:4569]

e[Kasterisk1*CLI>
– Executing e[1;36;40mGotoe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40ms-CHANUNAVAIL|1e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Goto (macro-vm,s-CHANUNAVAIL,1)

e[Kasterisk1*CLI>
– Executing e[1;36;40mMacroe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mget-vmcontext|4001e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mSete[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mVMCONTEXT=defaulte[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?200:300e[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Goto (macro-get-vmcontext,s,300)

e[Kasterisk1*CLI>
– Executing e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack

e[Kasterisk1*CLI>
– Executing e[1;36;40mVoiceMaile[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40mu4001@defaulte[0;37;40m”) in new stack

e[Kasterisk1*CLI>
We’re at 192.168.1.101 port 16746

e[Kasterisk1*CLI>
Adding codec 0x1 (g723) to SDP

e[Kasterisk1*CLI>
Adding codec 0x400 (ilbc) to SDP

e[Kasterisk1*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

e[Kasterisk1*CLI>
Reliably Transmitting (no NAT) to 192.168.1.102:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK6e920000fea9ffff;received=192.168.1.102

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone;tag=as5ca8d837

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18714 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: sip:4001@192.168.1.101

Content-Type: application/sdp

Content-Length: 242

v=0

o=root 2579 2579 IN IP4 192.168.1.101

s=session

c=IN IP4 192.168.1.101

t=0 0

m=audio 16746 RTP/AVP 4 97 101

a=rtpmap:4 G723/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


e[Kasterisk1*CLI>
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2000-80e4’ in macro ‘vm’

e[Kasterisk1*CLI>
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2000-80e4’ in macro ‘exten-vm’

e[Kasterisk1*CLI>
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2000-80e4’

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:
ACK sip:4001@192.168.1.101 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK5576000003b30000

From: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

To: sip:4001@192.168.1.101;user=phone;tag=as5ca8d837

Contact: sip:2000@192.168.1.102;user=phone

Proxy-Authorization: Digest username=“2000”, realm=“asterisk”, algorithm=MD5, uri="sip:4001@192.168.1.101", nonce=“23618af7”, response=“f2f945610697cfb81b49817f23333486”

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 18714 ACK

User-Agent: Grandstream HT496 1.0.2.16

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Content-Length: 0

e[Kasterisk1*CLI>

— (12 headers 0 lines)—

e[Kasterisk1*CLI>
set_destination: Parsing sip:2000@192.168.1.102;user=phone for address/port to send to
set_destination: set destination to 192.168.1.102, port 5060
Reliably Transmitting (no NAT) to 192.168.1.102:5060:
BYE sip:2000@192.168.1.102 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK4a5480f1

From: sip:4001@192.168.1.101;user=phone;tag=as5ca8d837

To: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

Contact: sip:4001@192.168.1.101

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 102 BYE

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0


e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK4a5480f1

From: sip:4001@192.168.1.101;user=phone;tag=as5ca8d837

To: sip:2000@192.168.1.101;user=phone;tag=2fbdfffff489ffff

Call-ID: 1e29ffff5b2bffff@192.168.1.102

CSeq: 102 BYE

User-Agent: Grandstream HT496 1.0.2.16

Contact: sip:2000@192.168.1.102;user=phone

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Supported: replaces

Content-Length: 0

— (11 headers 0 lines)—
Destroying call ‘1e29ffff5b2bffff@192.168.1.102’

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 00835 DCall: 00000 [192.168.1.100:4569]
USERNAME : 4001
REFRESH : 60

Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00004ms SCall: 00005 DCall: 00835 [192.168.1.100:4569]
AUTHMETHODS : 3
CHALLENGE : 113768729
USERNAME : 4001

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ
Timestamp: 00031ms SCall: 00835 DCall: 00005 [192.168.1.100:4569]
USERNAME : 4001
REFRESH : 60
MD5 RESULT : a05a553763eff5bf62eafb43c849d2d0

Tx-Frame Retry[000] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK
Timestamp: 00019ms SCall: 00005 DCall: 00835 [192.168.1.100:4569]
USERNAME : 4001
DATE TIME : 2006-04-26 13:36:14
REFRESH : 60
APPARENT ADDRES : IPV4 192.168.1.100:4569
MESSAGE COUNT : 0
CALLING NUMBER : 4001
CALLING NAME : IAXLaptop

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00019ms SCall: 00835 DCall: 00005 [192.168.1.100:4569]

e[Kasterisk1*CLI>

<-- SIP read from 192.168.1.102:5060:

— (0 headers 0 lines) Nat keepalive —

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 13277 DCall: 00000 [192.168.1.103:4569]
USERNAME : 4000
REFRESH : 60

Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00005ms SCall: 00001 DCall: 13277 [192.168.1.103:4569]
AUTHMETHODS : 3
CHALLENGE : 300801077
USERNAME : 4000

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 13277 DCall: 00001 [192.168.1.103:4569]
USERNAME : 4000
REFRESH : 60
MD5 RESULT : 6bd3818a8473da8e236fa57d2f4534ef

Tx-Frame Retry[000] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK
Timestamp: 00011ms SCall: 00001 DCall: 13277 [192.168.1.103:4569]
USERNAME : 4000
DATE TIME : 2006-04-26 13:36:32
REFRESH : 60
APPARENT ADDRES : IPV4 192.168.1.103:4569
MESSAGE COUNT : 0
CALLING NUMBER : 4000
CALLING NAME : IAX

e[Kasterisk1*CLI>
Rx-Frame Retry[ No] – OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00011ms SCall: 13277 DCall: 00001 [192.168.1.103:4569]

e[Kasterisk1*CLI>


#8

e[Kasterisk1*CLI>
– Hungup ‘IAX2/4001-2’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mSIP/2000-80e4e[0;37;40m”, “e[1;35;40m0?s-CHANUNAVAIL|1e[0;37;40m”) in new stack

why ???


#9

What IAX endpoint are you using?


#10

the whole system is only on lan with Linksys w router. IAX I am using idefisk softphones, SIP I have Grandstream 496 ATA adapt. and BOL SIPPhones softphones.

if I do IAX to SIP, it rings, but hangs up as soon as you try to aswer it
if I do SIP to IAX, no ring, nothing. Busy signal on ATA

But it works good SIP to SIP, or IAX to IAX


#11

Anybody has any idea?


#12

anybody at all?


#13

anybody???..