VoIP Mailing List Archives

[danikpro at gmail.com]

Here is the log, my extensions is in the default section

*CLI> core set verbose 3
Verbosity is at least 3
*CLI> [Mar 5 15:21:43] NOTICE[15870]: chan_sip.c:13879
handle_request_invite: Call from '7007' to extension '999' rejected
because extension not found.
*CLI> sip set debug
SIP Debugging enabled
*CLI>
<--- SIP read from 192.168.85.27:5060 --->
INVITE sip:999 at 192.168.85.29 SIP/2.0
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:7007 at 192.168.85.27:5060>
To: "999"<sip:999 at 192.168.85.29>
From: "dan"<sip:7007 at 192.168.85.29>;tag=4773d83f
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 371

v=0
o=- 7 2 IN IP4 192.168.85.27
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.85.27
t=0 0
m=audio 5062 RTP/AVP 107 119 100 106 0 105 98 8 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 15 lines) ---
Sending to 192.168.85.27 : 5060 (NAT)
Using INVITE request as basis request -
NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
Found user '7007'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.85.27:5062
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60c
(ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.85.27:5062
Looking for 999 in default (domain 192.168.85.29)

<--- Reliably Transmitting (no NAT) to 192.168.85.27:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;received=192.168.85.27;rport=5060
From: "dan"<sip:7007 at 192.168.85.29>;tag=4773d83f
To: "999"<sip:999 at 192.168.85.29>;tag=as2eff0a82
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Mar 5 15:22:17] NOTICE[15870]: chan_sip.c:13879
handle_request_invite: Call from '7007' to extension '999' rejected
because extension not found.
Scheduling destruction of SIP dialog
'NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.' in 32000 ms (Method:
INVITE)

<--- SIP read from 192.168.85.27:5060 --->
ACK sip:999 at 192.168.85.29 SIP/2.0
Via: SIP/2.0/UDP
192.168.85.27:5060;branch=z9hG4bK-d87543-35763506b9430b2b-1--d87543-;rport
To: "999"<sip:999 at 192.168.85.29>;tag=as2eff0a82
From: "dan"<sip:7007 at 192.168.85.29>;tag=4773d83f
Call-ID: NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'NjRjMmE1MDQzYjc5YjM3OTJiYmJkN2YzMGY2ZDMzNDQ.' Method: ACK
sip sedialplan show
[ Context 'ael-default' created by 'pbx_ael' ]
Include => 'ael-demo' [pbx_ael]

[ Context 'ael-demo' created by 'pbx_ael' ]
'#' => 1. Playback(demo-thanks) [pbx_ael]
2. Hangup() [pbx_ael]
'1000' => 1. Goto(ael-default|s|1) [pbx_ael]
'2' => 1. Background(demo-moreinfo) [pbx_ael]
2. Goto(s|instructions) [pbx_ael]
'3' => 1. Set(LANGUAGE()=fr) [pbx_ael]
2. Goto(s|restart) [pbx_ael]
'500' => 1. Playback(demo-abouttotry) [pbx_ael]
2. Dial(IAX2/guest at misery.digium.com/s at default) [pbx_ael]
3. Playback(demo-nogo) [pbx_ael]
4. Goto(s|instructions) [pbx_ael]
'600' => 1. Playback(demo-echotest) [pbx_ael]
2. Echo() [pbx_ael]
3. Playback(demo-echodone) [pbx_ael]
4. Goto(s|instructions) [pbx_ael]
'8500' => 1. VoicemailMain() [pbx_ael]
2. Goto(s|instructions) [pbx_ael]
'i' => 1. Playback(invalid) [pbx_ael]
's' => 1. Wait(1) [pbx_ael]
2. Answer() [pbx_ael]
3. Set(TIMEOUT(digit)=5) [pbx_ael]
4. Set(TIMEOUT(response)=10) [pbx_ael]
[restart] 5. Background(demo-congrats) [pbx_ael]
[instructions] 6. Set(x=$[0]) [pbx_ael]
7. GotoIf($[ ${x} < 3]?8:12) [pbx_ael]
8. Background(demo-instruct) [pbx_ael]
9. WaitExten() [pbx_ael]
10. Set(x=$[${x} + 1]) [pbx_ael]
11. Goto(7) [pbx_ael]
12. NoOp(Finish for-ael-demo-3) [pbx_ael]
't' => 1. Goto(#|1) [pbx_ael]
'_1234' => 1. Macro(ael-std-exten-ael|${EXTEN}| "IAX2") [pbx_ael]

[ Context 'macro-ael-std-exten-ael' created by 'pbx_ael' ]
'a' => 1. VoiceMailMain(${ext}) [pbx_ael]
2. Goto(3) [pbx_ael]
3. NoOp(End of Extension a) [pbx_ael]
's' => 1. Set(ext=${ARG1}) [pbx_ael]
2. Set(dev=${ARG2}) [pbx_ael]
3. Dial(${dev}/${ext}|20) [pbx_ael]
4. Goto(sw-1-${DIALSTATUS}|10) [pbx_ael]
5. NoOp(Finish switch-ael-std-exten-ael-1) [pbx_ael]
6. Goto(7) [pbx_ael]
7. NoOp(End of Macro ael-std-exten-ael-s) [pbx_ael]
'sw-1-' => 10. Goto(sw-1-.|10) [pbx_ael]
'sw-1-BUSY' => 10. Voicemail(${ext}|b) [pbx_ael]
11. Goto(s|5) [pbx_ael]
'_sw-1-.' => 10. Voicemail(${ext}|u) [pbx_ael]
11. Goto(s|5) [pbx_ael]

[ Context 'ael-local' created by 'pbx_ael' ]
Include => 'ael-default' [pbx_ael]
Include => 'ael-trunklocal' [pbx_ael]
Include => 'ael-iaxtel700' [pbx_ael]
Include => 'ael-trunktollfree' [pbx_ael]
Include => 'ael-iaxprovider' [pbx_ael]
Ignore pattern => '9' [pbx_ael]

[ Context 'ael-longdistance' created by 'pbx_ael' ]
Include => 'ael-local' [pbx_ael]
Include => 'ael-trunkld' [pbx_ael]
Ignore pattern => '9' [pbx_ael]

[ Context 'ael-international' created by 'pbx_ael' ]
Include => 'ael-longdistance' [pbx_ael]
Include => 'ael-trunkint' [pbx_ael]
Ignore pattern => '9' [pbx_ael]

[ Context 'ael-trunktollfree' created by 'pbx_ael' ]
'_91800NXXXXXX' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]
'_91866NXXXXXX' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]
'_91877NXXXXXX' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]
'_91888NXXXXXX' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]

[ Context 'ael-trunklocal' created by 'pbx_ael' ]
'_9NXXXXXX' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]

[ Context 'ael-trunkld' created by 'pbx_ael' ]
'_91NXXNXXXXXX' => 1. Macro(ael-dundi-e164|${EXTEN:1}) [pbx_ael]
2. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]
Include => 'ael-dundi-e164-lookup' [pbx_ael]

[ Context 'ael-trunkint' created by 'pbx_ael' ]
'_9011.' => 1. Macro(ael-dundi-e164|${EXTEN:4}) [pbx_ael]
2. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [pbx_ael]
Include => 'ael-dundi-e164-lookup' [pbx_ael]

[ Context 'ael-iaxprovider' created by 'pbx_ael' ]

[ Context 'ael-iaxtel700' created by 'pbx_ael' ]
'_91700XXXXXXX' => 1.
Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) [pbx_ael]

[ Context 'macro-ael-dundi-e164' created by 'pbx_ael' ]
's' => 1. Set(exten=${ARG1}) [pbx_ael]
2. Goto(${exten}|1) [pbx_ael]
3. Goto(4) [pbx_ael]
4. NoOp(End of Macro ael-dundi-e164-s) [pbx_ael]

[ Context 'ael-dundi-e164-lookup' created by 'pbx_ael' ]
Include => 'ael-dundi-e164-local' [pbx_ael]
Include => 'ael-dundi-e164-switch' [pbx_ael]

[ Context 'ael-dundi-e164-switch' created by 'pbx_ael' ]
Alt. Switch => 'DUNDi/e164' [pbx_ael]

[ Context 'ael-dundi-e164-local' created by 'pbx_ael' ]
Include => 'ael-dundi-e164-canonical' [pbx_ael]
Include => 'ael-dundi-e164-customers' [pbx_ael]
Include => 'ael-dundi-e164-via-pstn' [pbx_ael]

[ Context 'ael-dundi-e164-via-pstn' created by 'pbx_ael' ]

[ Context 'ael-dundi-e164-customers' created by 'pbx_ael' ]

[ Context 'ael-dundi-e164-canonical' created by 'pbx_ael' ]

[ Context 'default' created by 'pbx_config' ]
'7007' => 1. Goto(7007-${CNT}|1) [pbx_config]
'7007-2' => 1. Set(GLOBAL(CNT)=1) [pbx_config]
2. Answer() [pbx_config]
3. Playback(hello-world|skip) [pbx_config]
4. Hangup() [pbx_config]
'7008' => 1. Set(GLOBAL(connid)=0) [pbx_config]
2. Set(GLOBAL(resultid)=0) [pbx_config]
3. Set(GLOBAL(fetchid)=0) [pbx_config]
4. MYSQL(Connect connid localhost root test test)
[pbx_config]
5. MYSQL(Query resultid ${connid} Select a from a)
[pbx_config]
6. MYSQL(Fetch fetchid ${resultid} a) [pbx_config]
7. MYSQL(Clear ${resultid}) [pbx_config]
8. MYSQL(Disconnect ${connid}) [pbx_config]
9. goto(${a}|1) [pbx_config]
's' => 1. Answer() [pbx_config]
'_7007-.' => 1. Set(GLOBAL(CNT)=$[${CNT}+1]) [pbx_config]
2. Answer() [pbx_config]
3. Playback(tt-weasels|skip) [pbx_config]
4. Hangup() [pbx_config]

[ Context 'parkedcalls' created by 'res_features' ]
'700' => 1. Park() [res_features]

-= 31 extensions (81 priorities) in 21 contexts. =-
*CLI> Reliably Transmitting (no NAT) to 192.168.85.27:5060:
OPTIONS sip:192.168.85.27 SIP/2.0
Via: SIP/2.0/UDP 192.168.85.29:5060;branch=z9hG4bK4fdf43bb;rport
From: "asterisk" <sip:asterisk at 192.168.85.29>;tag=as338aef9f
To: <sip:192.168.85.27>
Contact: <sip:asterisk at 192.168.85.29>
Call-ID: 4dfa4d734386c9d40abd873e2c6125d3 at 192.168.85.29
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Mar 2008 11:23:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 192.168.85.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.85.29:5060;branch=z9hG4bK4fdf43bb;rport=5060
Contact: <sip:192.168.85.27:5060>
To: <sip:192.168.85.27>;tag=e4752e7d
From: "asterisk"<sip:asterisk at 192.168.85.29>;tag=as338aef9f
Call-ID: 4dfa4d734386c9d40abd873e2c6125d3 at 192.168.85.29
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog
'4dfa4d734386c9d40abd873e2c6125d3 at 192.168.85.29' Method: OPTIONS


2008/3/8, Tzafrir Cohen <tzafrir.cohen at xorcom.com>:

[tilghman at mail.jeffa...]

On Saturday 08 March 2008 02:52:39 Daniel Suleyman wrote:

[danikpro at gmail.com]

ok, then I'm not understanding something.
How I can call with xlite to my Asterisk not sending extension?
when I want to call I pick up phone, hear ring (piiiiiii) and I need
to type some extension otherwise nothing hapens....

2008/3/8, Tilghman Lesher <tilghman at mail.jeffandtilghman.com>:

[tilghman at mail.jeffa...]

On Sunday 09 March 2008 00:19:05 Daniel Suleyman wrote:

[tzafrir.cohen at xorco...]

On Sun, Mar 09, 2008 at 10:19:05AM +0400, Daniel Suleyman wrote:

[danikpro at gmail.com]

Thank you for guide most things become cleare. No I dont need the dial tone.
When I pickup XLITE to dial a number I hear dialtone and after I enter
number nothing happens, this behaviar was strange for me, exactly
becase you said I have analog phone in mind
2008/3/9, Tzafrir Cohen <tzafrir.cohen at xorcom.com>:

[noahisaacmiller at gma...]

Hi Daniel -

[danikpro at gmail.com]

thanks.

2008/3/10, Noah Miller <noahisaacmiller at gmail.com>: