Call from '200' to extension '8500' rejected because

Asterisk Asterisk Support
1

While editing extensions.conf withvi some glitch in vi had blown out my extentions.conf file. Totally mysterious. So I had to rewrite extentions.conf by hand.

I am getting the error of " Call from ‘200’ to extension ‘8500’ rejected because extension not found. I will post sip.conf and extensions.conf plus the error log.

SIP.conf

[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls

[199]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=199
secret=199
context=internal
regexten=199 ; When they register, create extension 1234
callerid=“guest” <199>
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
qualify=yes
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
mailbox=199@default ; Subscribe to status of multiple mailboxes

[200]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=200
secret=200
context=internal
regexten=200 ; When they register, create extension 1234
callerid=“JoeMonday” <200>
host=dynamic ; This device needs to register
nat=yes ; X-Lite is behind a NAT router
qualify=yes
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
mailbox=200@default ; Subscribe to status of multiple mailboxes

extentions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/G2 ; Trunk interface
TRUNKMSD=1

;--------------------------Custom Config for this asterisk box-------------------------

[default-custom]

;–Inbound dahdi trunk from--------
;–xxxxx CLEC provider. This provider info is not listed in this message body–

exten => s,1,Answer()
exten => s,2,Background(outgoing)
exten => s,3,DigitTimeout(5)
exten => s,4,ResponceTimeout(10)

exten => 1,1,Goto(internal,200,1)
exten => 2,1,Goto(internal,201,1)
exten => 3,1,Goto(conferance,1,1)

exten => i,1,Answer()
exten => i,2,Playback(pbx-invalid)
exten => i,3,Goto(s,2)

[record-outgoing-msg]

exten => 8501,1,Playback(recorder-record-after-beep)
exten => 8501,2,Record(/var/lib/asterisk/sounds/outgoing:wav,3,30)
exten => 8501,3,Playback(record-menu) ;Press 1 to Accept, Press 2 to review, Press 3 to re-record
exten => 8501,4,WaitExten()

exten => 2,1,Playback(outgoing)
exten => 2,n,Goto(8501,3)

exten => 3,1,Record(/var/lib/asterisk/sounds/outgoing:wav,3,30)
exten => 3,n,Playback(outgoing)
exten => 3,n,Goto(8501,3)

exten => 1,1,Playback(goodbye)
exten => 1,n,Hangup()

[dahditrunk]
exten => _NXXNXXXXXX,1,Dial(dahdi/1/${EXTEN})
exten => _1NXXNXXXXXX,1,Dial(dahdi/1/${EXTEN})
exten => _1800XXXXXXX,1,Dial(dahdi/1/${EXTEN})
exten => _1888XXXXXXX,1,Dial(dahdi/1/${EXTEN})
exten => 0,1,Dial(dahdi/1/${EXTEN})

exten => 411,1,Dial(dahdi/1/${EXTEN})
exten => 411,n,Hangup()

exten => 911,1,Dial(dahdi/1/${EXTEN})
exten => 911,n,Hangup()

[voicemailmain]
exten => 8500,1,Answer()
exten => 8500,n,VoicemailMain
exten => 8500,n,Hangup

[internal]
include => default-custom
include => voicemailmain
include => record-outgoing-msg
include => dahditrunk

exten => 0607,1,Dial(SIP/${EXTEN},20,Tt)
exten => 0607,n,VoiceMail(u200@default)
exten => 0607,n,VoiceMail(b200@default)
exten => 0607,n,Hangup()

exten => 199,1,Dial(SIP/${EXTEN},20,Tt)
exten => 199,2,VoiceMail(u200@default)
exten => 199,102,VoiceMail(b200@default)
exten => 199,103,Hangup()

exten => 200,1,Dial(SIP/${EXTEN},20,Tt)
exten => 200,2,VoiceMail(u200@default)
exten => 200,102,VoiceMail(b200@default)
exten => 200,103,Hangup()

/var/log/asterisk/message

[Feb 14 17:49:50] WARNING[3673] chan_mgcp.c: Unable to get our IP address, MGCP disabled
[Feb 14 17:49:50] WARNING[3670] chan_sip.c: Format for authentication entry is user[:secret]@realm at line 683
[Feb 14 17:49:55] NOTICE[3670] chan_sip.c: Call from ‘200’ to extension ‘8500’ rejected because extension not found.
[Feb 14 17:49:58] NOTICE[3670] chan_sip.c: Call from ‘200’ to extension ‘199’ rejected because extension not found.
[Feb 14 18:04:29] NOTICE[3827] cdr.c: CDR simple logging enabled.
[Feb 14 18:04:29] NOTICE[3827] indications.c: Removed default indication country ‘us’
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: Starting AEL load process.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: calculated config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] pbx_ael.c: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
[Feb 14 18:04:29] NOTICE[3827] app_playback.c: Reloading say.conf
[Feb 14 18:04:29] WARNING[3827] pbx_dundi.c: Unable to look up host ‘ApturaPBX’
[Feb 14 18:04:29] WARNING[3827] config.c: Unterminated comment detected beginning on line 628
[Feb 14 18:04:29] WARNING[3673] chan_mgcp.c: Unable to get our IP address, MGCP disabled
[Feb 14 18:04:29] WARNING[3827] pbx.c: Context ‘default-custom’ tries to include nonexistent context ‘voicemailman’
[Feb 14 18:04:29] WARNING[3827] chan_dahdi.c: Ignoring switchtype
[Feb 14 18:04:29] WARNING[3827] chan_dahdi.c: Ignoring signalling
[Feb 14 18:04:29] WARNING[3827] chan_dahdi.c: Ignoring rxwink
[Feb 14 18:04:29] WARNING[3670] chan_sip.c: Format for authentication entry is user[:secret]@realm at line 683
[Feb 14 18:04:32] NOTICE[3670] chan_sip.c: Call from ‘200’ to extension ‘8500’ rejected because extension not found.
[Feb 14 18:45:20] NOTICE[3670] chan_sip.c: Call from ‘200’ to extension ‘8501’ rejected because extension not found.

As you can see, extention 8500 is rejected from 200 as well as 8501 is rejcted from 200

CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status
201/201 192.168.10.20 D 5060 OK (17 ms)
200/200 192.168.10.50 D N 5060 OK (52 ms)

CLI>module show

res_musiconhold.so Music On Hold Resource 0
res_agi.so Asterisk Gateway Interface (AGI) 0
res_crypto.so Cryptographic Digital Signatures 0
res_speech.so Generic Speech Recognition API 0
res_indications.so Indications Resource 0
res_monitor.so Call Monitoring Resource 0
res_adsi.so ADSI Resource 0
res_features.so Call Features Resource 0
res_smdi.so Simplified Message Desk Interface (SMDI) 0
app_url.so Send URL Applications 0
app_controlplayback.so Control Playback Application 0
app_externalivr.so External IVR Interface Application 0
app_channelredirect.so Channel Redirect 0
app_ices.so Encode and Stream via icecast and ices 0
func_cut.so Cut out information from a string 0
res_clioriginate.so Call origination from the CLI 0
app_dahdiscan.so Scan Zap channels application 0
func_timeout.so Channel timeout dialplan functions 0
app_mixmonitor.so Mixed Audio Monitoring Application 0
chan_phone.so Linux Telephony API Support 0
chan_agent.so Agent Proxy Channel 0
app_directory.so Extension Directory 0
app_dahdiras.so DAHDI RAS Application 0
format_h264.so Raw H.264 data 0
pbx_ael.so Asterisk Extension Language Compiler 0
codec_g726.so ITU G.726-32kbps G726 Transcoder 0
app_verbose.so Send verbose output 0
app_setcdruserfield.so CDR user field apps 0
app_dahdibarge.so Barge in on channel application 0
app_readfile.so Stores output of file into a variable 0
app_hasnewvoicemail.so Indicator for whether a voice mailbox ha 0
app_dial.so Dialing Application 0
app_macro.so Extension Macros 0
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0
pbx_loopback.so Loopback Switch 0
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0
app_flash.so Flash channel application 0
app_disa.so DISA (Direct Inward System Access) Appli 0
cdr_manager.so Asterisk Manager Interface CDR Backend 0
app_playback.so Sound File Playback Application 0
app_alarmreceiver.so Alarm Receiver for Asterisk 0
app_zapateller.so Block Telemarketers with Special Informa 0
codec_ulaw.so mu-Law Coder/Decoder 0
app_queue.so True Call Queueing 0
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0
format_jpeg.so JPEG (Joint Picture Experts Group) Image 0
pbx_dundi.so Distributed Universal Number Discovery ( 0
func_groupcount.so Channel group dialplan functions 0
func_db.so Database (astdb) related dialplan functi 0
pbx_realtime.so Realtime Switch 0
func_math.so Mathematical dialplan function 0
app_morsecode.so Morse code 0
codec_gsm.so GSM Coder/Decoder 0
app_db.so Database Access Functions 0
func_rand.so Random number dialplan function 0
app_directed_pickup.so Directed Call Pickup Application 0
func_global.so Global variable dialplan functions 0
app_dumpchan.so Dump Info About The Calling Channel 0
format_sln.so Raw Signed Linear Audio support (SLN) 0
app_chanspy.so Listen to the audio of an active channel 0
func_logic.so Logical dialplan functions 0
app_adsiprog.so Asterisk ADSI Programming Application 0
format_gsm.so Raw GSM data 0
func_strings.so String handling dialplan functions 0
pbx_spool.so Outgoing Spool Support 0
cdr_custom.so Customizable Comma Separated Values CDR 0
chan_local.so Local Proxy Channel (Note: used internal 0
app_image.so Image Transmission Application 0
app_lookupcidname.so Look up CallerID Name from local databas 0
app_settransfercapability.so Set ISDN Transfer Capability 0
app_waitforsilence.so Wait For Silence 0
chan_sip.so Session Initiation Protocol (SIP) 0
chan_skinny.so Skinny Client Control Protocol (Skinny) 0
func_channel.so Channel information dialplan function 0
func_enum.so ENUM related dialplan functions 0
app_talkdetect.so Playback with Talk Detection 0
format_ilbc.so Raw iLBC data 0
app_exec.so Executes dialplan applications 0
format_vox.so Dialogic VOX (ADPCM) File Format 0
app_page.so Page Multiple Phones 0
app_sayunixtime.so Say time 0
func_base64.so base64 encode/decode dialplan functions 0
format_g729.so Raw G729 data 0
app_voicemail.so Comedian Mail (Voicemail System) 0
func_language.so Channel language dialplan function 0
app_authenticate.so Authentication Application 0
app_softhangup.so Hangs up the requested channel 0
app_transfer.so Transfer 0
chan_mgcp.so Media Gateway Control Protocol (MGCP) 0
func_md5.so MD5 digest dialplan functions 0
cdr_csv.so Comma Separated Values CDR Backend 0
app_followme.so Find-Me/Follow-Me Application 0
func_realtime.so Read/Write values from a RealTime reposi 0
app_meetme.so MeetMe conference bridge 0
codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0
app_forkcdr.so Fork The CDR into 2 separate entities 0
app_privacy.so Require phone number to be entered, if n 0
app_dictate.so Virtual Dictation Machine 0
app_userevent.so Custom User Event Application 0
codec_alaw.so A-law Coder/Decoder 0
app_festival.so Simple Festival Interface 0
app_getcpeid.so Get ADSI CPE ID 0
app_waitforring.so Waits until first ring after time 0
res_convert.so File format conversion CLI command 0
app_speech_utils.so Dialplan Speech Applications 0
app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0
app_read.so Read Variable Application 0
pbx_config.so Text Extension Configuration 0
format_h263.so Raw H.263 data 0
app_mp3.so Silly MP3 Application 0
chan_dahdi.so DAHDI Telephony 0
chan_oss.so OSS Console Channel Driver 0
func_callerid.so Caller ID related dialplan function 0
app_parkandannounce.so Call Parking and Announce Application 0
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0
app_record.so Trivial Record Application 0
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0
func_moh.so Music-on-hold dialplan function 0
app_stack.so Stack Routines 0
format_g726.so Raw G.726 (16/24/32/40kbps) data 0
func_uri.so URI encode/decode dialplan functions 0
format_g723.so G.723.1 Simple Timestamp File Format 0
app_amd.so Answering Machine Detection Application 0
app_nbscat.so Silly NBS Stream Application 0
app_lookupblacklist.so Look up Caller*ID name/number from black 0
app_while.so While Loops and Conditional Execution 0
app_test.so Interface Test Application 0
app_setcallerid.so Set CallerID Application 0
format_wav.so Microsoft WAV format (8000Hz Signed Line 0
app_chanisavail.so Check channel availability 0
app_realtime.so Realtime Data Lookup/Rewrite 0
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0
app_cdr.so Tell Asterisk to not maintain a CDR for 0
app_sms.so SMS/PSTN handler 0
func_env.so Environment/filesystem dialplan function 0
app_echo.so Simple Echo Application 0
app_sendtext.so Send Text Applications 0
func_sha1.so SHA-1 computation dialplan function 0
app_system.so Generic System() application 0
func_cdr.so CDR dialplan function 0
app_senddtmf.so Send DTMF digits Application 0
app_random.so Random goto 0
142 modules loaded

SO NOW im at a cross roads. What else could be the problem? I am only testing for the internal calls phone to phone and phone to vm at this point. Not concerned with external dahdi or sip accounts.

Thanks

2

What happens when you include -> internal in voicemailmail?

3

I see a few potential problems in the log: -

[quote][Feb 14 18:04:29] WARNING[3827] config.c: Unterminated comment detected beginning on line 628 [/quote]That looks quite serious though I cant see what config file it’s in, it may cause something to be ignored.

[quote][Feb 14 18:04:29] WARNING[3827] pbx.c: Context ‘default-custom’ tries to include nonexistent context ‘voicemailman’ [/quote]There’s a typo in an include somewhere voicemailman should be voicemailmain.

[quote][Feb 14 18:04:29] WARNING[3670] chan_sip.c: Format for authentication entry is user[:secret]@realm at line 683 [/quote]May be worth looking at in sip.conf but may have nothing to do with your issue.

Try turning up the verbosity (core set verbose 4) on the CLI and watching the CLI as the call is attempted. Something MUST be in the wrong context.

4

Some times you can stare at something untill you are blue in the face and not relize your mistake. I will check the typos again then get back to you all. Also, will get that line from chan.c and put it here.

Lastly, I didnt do to bad making this almost all by memory but wont know untill I test it all.

So this config file disapearence does not happen again what do most of you do for backup purpouses? shove the config files in a customer account name file directory on your companies ftp server?
Idealy, I would want to have a full running version of linux and asterisk with configs in a ftp account or usb drive to get a future customer pbx up and running in under a hour.

Post your ideas if posible. I could make this a new thread to.

5

[default-custom]
;include => trunktollfree
;include => local
;include => demo
;include => default
include => voicemailman -< oops!

with this error for incoming calls, it did not change the behavior of voicemailmain as a include in [internal] for my sip phones. Look below

[voicemailmain]
include => internal
exten => 8500,1,Answer()
exten => 8500,n,VoicemailMain
exten => 8500,n,Hangup

;---------------------------------------------------
[internal]
include => default-custom
include => inbound ;vitel inbound DID and outbound
include => conferance
include => voicemailmain
include => dahditrunk
include => record-outgoing-msg

exten => 0607,1,Dial(SIP/${EXTEN},20,Tt)
exten => 0607,n,VoiceMail(u200@default)
exten => 0607,n,VoiceMail(b200@default)
exten => 0607,n,Hangup()

exten => 199,1,Dial(SIP/${EXTEN},20,Tt)
exten => 199,2,VoiceMail(u200@default)
exten => 199,102,VoiceMail(b200@default)
exten => 199,103,Hangup()

exten => 200,1,Dial(SIP/${EXTEN},20,Tt)
exten => 200,2,VoiceMail(u200@default)
exten => 200,102,VoiceMail(b200@default)
exten => 200,103,Hangup()

Here is my error list in /var/log/asterisk/message tail -20

[voicemailmain] <-- spelled correctly
include => internal
exten => 8500,1,Answer()
exten => 8500,n,VoicemailMain
exten => 8500,n,Hangup

;---------------------------------------------------
[internal]
include => default-custom
include => inbound ;vitel inbound DID and outbound
include => conferance
include => voicemailmain <-- spelled correctly
include => dahditrunk
include => record-outgoing-msg

;— house sip and polycom phones—

exten => 0607,1,Dial(SIP/${EXTEN},20,Tt)
exten => 0607,n,VoiceMail(u200@default)
exten => 0607,n,VoiceMail(b200@default)
exten => 0607,n,Hangup()

exten => 199,1,Dial(SIP/${EXTEN},20,Tt)
exten => 199,2,VoiceMail(u200@default)
exten => 199,102,VoiceMail(b200@default)
exten => 199,103,Hangup()

exten => 200,1,Dial(SIP/${EXTEN},20,Tt)
exten => 200,2,VoiceMail(u200@default)
exten => 200,102,VoiceMail(b200@default)
exten => 200,103,Hangup()

[Feb 15 11:41:29] WARNING[2983] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: Starting AEL load process.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: calculated config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] NOTICE[2983] pbx_ael.c: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
[Feb 15 11:41:30] WARNING[2983] pbx_dundi.c: Unable to look up host ‘ApturaPBX’
[Feb 15 11:41:31] WARNING[2983] chan_sip.c: Format for authentication entry is user[:secret]@realm at line 683
[Feb 15 11:41:31] WARNING[2983] chan_sip.c: insecure=very at line 704 is deprecated; use insecure=port,invite instead
[Feb 15 11:41:31] WARNING[2983] chan_skinny.c: Unable to get our IP address, Skinny disabled
[Feb 15 11:41:31] WARNING[2983] chan_mgcp.c: Unable to get our IP address, MGCP disabled
[Feb 15 11:41:31] ERROR[2983] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[Feb 15 11:41:31] NOTICE[3688] chan_sip.c: Peer ‘201’ is now Reachable. (123ms / 2000ms)
[Feb 15 11:41:31] NOTICE[3688] chan_sip.c: Peer ‘200’ is now Reachable. (59ms / 2000ms)
[Feb 15 11:41:31] WARNING[2983] config.c: Unterminated comment detected beginning on line 628
[Feb 15 11:41:31] WARNING[2983] pbx.c: Context ‘default-custom’ tries to include nonexistent context ‘voicemailman’
[Feb 15 11:41:31] NOTICE[3688] chan_sip.c: Peer ‘0607’ is now Reachable. (62ms / 2000ms)
[Feb 15 12:46:15] NOTICE[3688] chan_sip.c: Call from ‘200’ to extens

made test call after reload

[Feb 15 13:19:31] NOTICE[3688]: chan_sip.c:14383 handle_request_invite: Call from ‘200’ to extension ‘8500’ rejected because extension not found.

Could it be that there are so many issues with chan_sip.c that is it not able to read the extensions or the context or includes??

MY asterisk version is Asterisk 1.4.22 currently running on ApturaPBX

Here are the line errors on chan_sip.c located in /usr/src/asterisk-1.4.22/asterisk-1.4.22/channels

line 704 - AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */

line 683 blank and it is showing a error in messages?

   682 };

-> 683
684 /*! \brief Domain data structure.
685 \note In the future, we will connect this to a configuration tree specific
686 for this domain
687 /
688 struct domain {
689 char domain[MAXHOSTNAMELEN]; /!< SIP domain we are
responsible for /
690 char context[AST_MAX_EXTENSION]; /!< Incoming context for this
domain /
691 enum domain_mode mode; /!< How did we find this domain? /
692 AST_LIST_ENTRY(domain) list; /!< List mechanics */
693 };

config.c: Unterminated comment detected beginning on line 628

Not shure where this file is.

grep -r config.c /usr/src/asterisk-1.4.22/asterisk-1.4.22 comes up with a large number of possibilities with that work in it.

even after I changed the spelling of [voicemailmain] still did not make a difference and it should not have since this include was under the inomming dahdi context and not the [internal] context.

Thanks for your help in analyzing this issue. Want to get this system up as soon as possible.

6

Whoa! Hold on, I wasn’t quite clear here. I think that the line: [quote][Feb 15 11:41:31] WARNING[2983] config.c: Unterminated comment detected beginning on line 628[/quote]…refers to line 628 in one of your config files NOT in the c code.

I’d guess it’s in extensions.conf (or an include off of it) not sip.conf. Do a dialplan reload to test it and see if that triggers the same error.

7

[quote=“JamesK”]So this config file disapearence does not happen again what do most of you do for backup purpouses? shove the config files in a customer account name file directory on your companies ftp server?
Idealy, I would want to have a full running version of linux and asterisk with configs in a ftp account or usb drive to get a future customer pbx up and running in under a hour. [/quote]
Personally I use rsync in scripts / cron to maintain daily & / or weekly remote replicas of essential data (/etc/…) and user data. Basically if its user data or it took me more than 2 mins to customise then I rsync it somewhere and usually 2 places.

Oh and I keep a local replica on the server as well by the same method.

8

Okay, I will do a reload but I have already done it once. As far as the config file, where would the errors come from? the errors are pointing to the .c files not the config files.

I just reloaded and still getting the same error. I even put in the exten => 8500,1,VoiceMailMain in [internal] context where the phone extentions are listed.

I will look into rsync as I have never used it before.

9

Let’s take this message, purely as an example to illustrate: -

[quote][Feb 15 11:41:31] WARNING[2983] chan_sip.c: insecure=very at line 704 is deprecated; use insecure=port,invite instead[/quote]This means that the module calling itself “chan_sip.c” reported a problem (reading a config file) and the problem (in the config file) was “insecure=very at line 704 is deprecated; use insecure=port,invite instead”. The “line 704” refers to line 704 in /etc/asterisk/sip.conf.

How do I know it’s “sip.conf”? Well I have to guess that from the rest of the message!

Now, you have one reported by config.c. What I think’s going on there is that config.c is used to parse all config files for syntax, quotes, commenting, contexts etc., all the common stuff. During the parsing of one file (probably extensions.conf but could be another one) it’s reporting detected beginning on line 628". What exactly that means I’m not sure but the prog obviously considers it to be a problem.

It cannot be an “Unterminated comment” within the C code because it would never have compiled and asterisk would not be running.

10

I know that this is a apart of there configuration but right now, it is the least of my concerns. What I need to know is why one phone extention cannot ring another and CLI says that it cannot find the extention. I think at this point, I may have to do a fresh install even though that is not what I had in mind.

There is no reason why one extention cannot contact another if thay are in the same context and there respective sip accounts point to that context. Further, with both phones registered there still should be no reason why thay should not ring each other.

One thing I never mentioned in this original thread, about five days ago, I was in the middle of a sip call and CLI started to display repeating lines of messages saying that the password for x extention is invalid. What is really strange, is it would repeat this line over and over but change the exention to one that did not exist. It would show this in sequence like 101 password does not match, 102, password does not match. At least 50 lines if not more displayed this error with incremental sip.channels. extentions

11

And there you have it. There is no reason so * must THINK that they are NOT in the same context. Just thought - “dialplan show internal” would prove that one way or other. Other than that I’m lost - for the first time with this sort of fault.

Sounds like a brute force attempt to login from the public net. Is the server (or port 5060) exposed to the public net?

12

sip set debug off
SIP Debugging Disabled
ApturaPBXCLI> dialplan show internal
There is no existence of ‘internal’ context <— woo
ApturaPBXCLI>

grep [internal] /etc/asterisk/extentions.conf

[internal] <-- does not look missing to me!
include => default-custom
include => inbound ;vitel inbound DID and outbound
include => conferance
include => voicemailmain
include => dahditrunk
rest of internal continues from here…

For for those who may be asterisk developers, what part of asterisk would not read the context when it is visably present?

13

Now we’re getting there…

do this, to show all the contexts in extensions.conf: -

grep ‘^[’ /etc/asterisk/extensions.conf

Do you have other files included (#include s)? You’d need to grep them too.

now, in asterisk CLI, do: -

dialplan show ?

now compare the list. My guess is that the “dialplan show ?” context list will stop short (taking into account the fact that it’s not in the same order as the conf file) somewhere. Wherever it stops is approx where you have a MAJOR syntax issue in extensions.conf which is preventing it loading the rest of the file. Hence the missing contexts.

Of course that’s just my theory.

14

Jedi,

I did a dial plan show last night after thinking it is something I did not do. As expected, [internal] was not in the list of of other context. Now that I have identified that is we, have identified the issue, I am going to change the context to a different name and reload the dialplan and hopfully it shows up with show dialplan command. If it does, then I can change the context info in the extensions in sip.conf and my asterisk system should be up running properly.

15

Seems for some odd reason, pbx_config was not including my [internal] dial plan for my office phones into its configuration. I changed the name of internal to [internal-office] and changed context=internal-office on each of the phones in sip.conf extensions to match. This alone still did not affect the behavior of pbx_config after verifying my settings in dial plan show in CLI. So, I used a existing asterisk extension to test my phones dial plan configuration. I copied all my phones extension dial plan and placed it under [local]. I commented out the rest of local just for testing. I also commented out all of [internal-office]

Reloaded the dial plan and verified that my phones extensions were in fact loaded under [local]. All the extensions were loaded under local and made my test call. Well, my call went though.

So the question is, why is pbx_config being selective in what context it wants to include into the dial plan? If no dev comes forward this will continue with the owner of this section of asterisk.

16

Solved it!!!

The lines: -

[quote];–Inbound dahdi trunk from--------
;–xxxxx CLEC provider. This provider info is not listed in this message body–[/quote]
Cause the whole of the rest of the file to be skipped!

Why? Because “;–” (exactly 2 dashes) is the start of a multi-line comment and a “–;” is expected to end it. Without that end-comment the rest of extensions.conf is effectively commented out. Hence the message: -

…was REALLY very important.

Why didn’t I tell you that before? I never knew there were multi-line comments!

How’d I find it? I put your dialplan on my server then I looked up the source code & bingo. It’s real strange what you find sometimes.

Moral? Don’t start comments with “;–”.

17

I do not even recall on rules where you can place a comment at. But you are right in this case, as soon as I removed it the dial plan works. I can see there has been alot of interest in this thread which is good. It was a tough issue to troubleshoot. Perhaps there should be a note to not include comments after a context? or make future code ignore comments under the context. Now I can test the rest of this dial plan.

This thread is now closed…