What's wrong in my asterisk configuration?

gregchelli · November 13, 2014, 9:55pm

Hi, first sorry if i do some english error , its not my natif language , i have a Voip account with a provider (Poivy(i paid that 13$)) and i want to use asterisk with this account for all supplementary option of asterisk (like callerID). I dont want to use asterisk as a client , i use Zoiper free as sip/iax client software.

I installed asterisk successfully in my VM workstation’s Kali linux (debian 7) session,
i configured iax.conf , sip.conf and extensions.conf like this :

sip.conf :

[9001] type=peer host=dynamic username=900 secret=1234 context=outgoing

iax.conf :

[Poivy] type=peer host=sip.poivy.com username=My poivy username secret=My poivy password

Extensions :

[outgoing] exten => _1NXXNXXXXXX,1,SetCallerID(2024561111) exten => _1NXXNXXXXXX,n,Dial(IAX2/Poivy/${EXTEN})

when i log in with Zoiper free with my 9001/1234 sip account its registered but i cant make any call
Asterisk console when i try to make a outcall (I reload config) :

[code]root@gregchelli:~# asterisk -r
Asterisk 11.14.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.14.0 currently running on gregchelli (pid = 3221)
gregchelliCLI> reload
[Nov 13 22:46:04] NOTICE[5990]: app_queue.c:7822 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
[Nov 13 22:46:04] NOTICE[5990]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Nov 13 22:46:04] NOTICE[5990]: chan_skinny.c:7736 config_load: Configuring skinny from skinny.conf
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Nov 13 22:46:04] NOTICE[5990]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
gregchelliCLI> iax2 reload
[Nov 13 22:46:19] NOTICE[3389]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001
[Nov 13 22:46:19] NOTICE[3389]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001
[Nov 13 22:46:33] NOTICE[3389][C-00000003]: chan_sip.c:25649 handle_request_invite: Call from ‘900’ (192.168.1.28:5060) to extension ‘6176611901’ rejected because extension not found in context ‘outgoing’.
gregchelli*CLI>
[/code]

david55 · November 13, 2014, 11:24pm

6176611901 is not an 11 digit number beginning with 1 (although its second and fifth digits are non-zero).

Your Asterisk dialplan only matches NANP, national numbers, using the default long distance carrier.

david55 · November 13, 2014, 11:26pm

If English is not your native language, I wonder why you are trying to use North American style numbers. Most other countries use 0 as their national number prefix.

gregchelli · November 13, 2014, 11:38pm

Thanks a lot for your reply !

Yes im french
But my sip provider allow me to call all the world so i was belive that donest matter (probably a idiot thing)

Can you explain what i should change for using my configuration to call french number ?

In france international prefix is +33 and nation is 0
exemple : 0613305968

french number are 10 number and us 11 , so i should edit extensions.conf ?

thanks!

seer_m · November 14, 2014, 7:05am

If you want to call anywhere in the world (though international call charges may apply, you might want to ask your provider about that), your dialplan should have “exten => _.,n,Dial(IAX2/Poivy/${EXTEN})”. This takes any length, everything you punch into your phone.

Alternatively, you could use “exten => _0.,n,Dial(IAX2/Poivy/${EXTEN:1})”, which needs you to start dialing with a 0, then dial the actual number. “EXTEN:1” cuts out the 0 at the start. This is useful if you want to have more than one phone registered to your Asterisk server and be able to make calls between them without going through your provider.

Also, you can set your Caller ID in sip.conf, so your dialplan will be that one crucial line shorter.

david55 · November 14, 2014, 9:02am

_. is unsafe because it also matches the special extension like h.

gregchelli · November 14, 2014, 3:40pm

Hi, again thanks a lot for you help.

i changed my extensions.conf as you said but still dont work…
I want to put my callerID in sip.conf but dont know how i can do it
can you give me the option ? thanks

Asterisk console :

[code]root@gregchelli:~# asterisk -r
Asterisk 11.14.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.14.0 currently running on gregchelli (pid = 3221)
gregchelliCLI> reload
[Nov 14 16:31:33] NOTICE[4051]: app_queue.c:7822 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
[Nov 14 16:31:33] NOTICE[4051]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Nov 14 16:31:33] NOTICE[4051]: chan_skinny.c:7736 config_load: Configuring skinny from skinny.conf
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Nov 14 16:31:33] NOTICE[4051]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
[Nov 14 16:31:33] WARNING[4051]: pbx_config.c:1640 pbx_load_config: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘X.’ instead at line 860 of extensions.conf
[Nov 14 16:31:33] WARNING[4051]: pbx_config.c:1640 pbx_load_config: The use of '.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 861 of extensions.conf
[Nov 14 16:32:06] NOTICE[3381]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001
[Nov 14 16:32:06] NOTICE[3381]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001
[Nov 14 16:32:43] WARNING[4052][C-00000000]: pbx.c:4869 pbx_extension_helper: No application ‘SetCallerID’ for extension (outgoing, 0602690136, 1)
[Nov 14 16:32:43] WARNING[4052][C-00000000]: pbx.c:4869 pbx_extension_helper: No application ‘SetCallerID’ for extension (outgoing, h, 1)
gregchelliCLI>
[/code]

extension.conf :

[outgoing] exten => _.,1,SetCallerID(0626902626) exten => _.,n,Dial(IAX2/Poivy/${EXTEN})

I tried to delete CallId line to test but still dont work

david55 · November 14, 2014, 4:41pm

As it says, there is no SetCallerID application. You do it using Set(CALLERID(num)=…)

gregchelli · November 14, 2014, 5:20pm

Hi thanks for your last reply ! Yes david55 , im sorry im a beginner with asterisk and linux in general…

i changed extensions.conf as you said now :

extensions.conf :

[outgoing] exten => _.,1,Set(CALLERID(0626902626)=9001) exten => _.,n,Dial(IAX2/Poivy/${EXTEN})

console error :

Connected to Asterisk 11.14.0 currently running on gregchelli (pid = 3179) gregchelli*CLI> reload [Nov 14 18:15:04] NOTICE[4616]: app_queue.c:7822 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. [Nov 14 18:15:04] NOTICE[4616]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Nov 14 18:15:04] NOTICE[4616]: chan_skinny.c:7736 config_load: Configuring skinny from skinny.conf [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:164 pbx_load_module: Starting AEL load process. [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Nov 14 18:15:04] NOTICE[4616]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Nov 14 18:15:04] WARNING[4616]: pbx_config.c:1640 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 860 of extensions.conf [Nov 14 18:15:04] WARNING[4616]: pbx_config.c:1640 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 861 of extensions.conf [Nov 14 18:15:15] NOTICE[3409]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001 [Nov 14 18:15:15] NOTICE[3409]: chan_sip.c:27871 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 9001 [Nov 14 18:15:18] ERROR[4617][C-00000003]: func_callerid.c:1283 callerid_write: Unknown callerid data type '0626902626'. [Nov 14 18:15:22] NOTICE[3396]: chan_iax2.c:4830 __auto_congest: Auto-congesting call due to slow response [Nov 14 18:15:22] ERROR[4617][C-00000003]: func_callerid.c:1283 callerid_write: Unknown callerid data type '0626902626'. gregchelli*CLI>

david55 · November 14, 2014, 5:39pm

[Nov 14 18:15:18] ERROR[4617][C-00000003]: func_callerid.c:1283 callerid_write: Unknown callerid data type ‘0626902626’.

Please read the information I provided more carefully.

Note there should be built in help if you type:

core show function CALLERID

and

core show application set

at the CLI prompt.

gregchelli · November 14, 2014, 6:22pm

Yea i see i was so dumb sorry

i modify that with the right application form

extensions.conf :

[outgoing] exten => _.,1,Set(CALLERID(num)=0626902626) exten => _.,n,Dial(IAX2/Poivy/${EXTEN})

but still do not work

console :

Connected to Asterisk 11.14.0 currently running on gregchelli (pid = 3179) gregchelli*CLI> reload [Nov 14 19:20:10] NOTICE[7439]: app_queue.c:7822 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. [Nov 14 19:20:10] NOTICE[7439]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Nov 14 19:20:10] NOTICE[7439]: chan_skinny.c:7736 config_load: Configuring skinny from skinny.conf [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:164 pbx_load_module: Starting AEL load process. [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Nov 14 19:20:10] NOTICE[7439]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Nov 14 19:20:10] WARNING[7439]: pbx_config.c:1640 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 860 of extensions.conf [Nov 14 19:20:10] WARNING[7439]: pbx_config.c:1640 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 861 of extensions.conf [Nov 14 19:20:28] NOTICE[3395]: chan_iax2.c:4830 __auto_congest: Auto-congesting call due to slow response gregchelli*CLI>

What i already try : Put off all firewall in my computer and my router

david55 · November 14, 2014, 11:21pm

Turn up the verbosity and debug levels. You are only getting the final summary message, not the detailed reasons behind it.

gregchelli · November 15, 2014, 10:20am

Ok i turn up verbosity and debug file :
debug:

[quote][Nov 15 11:04:20] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:04:22] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:11] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:15] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:19] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:23] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC[/quote]
messages :

[quote][Nov 15 11:08:15] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:15] VERBOSE[6459] config.c: == Parsing ‘/etc/asterisk/logger.conf’: Found
[Nov 15 11:08:15] VERBOSE[6459] logger.c: Asterisk Queue Logger restarted
[Nov 15 11:08:19] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:19] VERBOSE[6459] config.c: == Parsing ‘/etc/asterisk/logger.conf’: Found
[Nov 15 11:08:19] VERBOSE[6459] logger.c: Asterisk Queue Logger restarted
[Nov 15 11:08:23] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 11:08:23] VERBOSE[6459] config.c: == Parsing ‘/etc/asterisk/logger.conf’: Found
[Nov 15 11:08:23] VERBOSE[6459] logger.c: Asterisk Queue Logger restarted
[Nov 15 11:08:41] VERBOSE[3528][C-00000014] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 15 11:08:41] VERBOSE[6460][C-00000014] pbx.c: – Executing [01914265847@outgoing:1] Set(“SIP/9001-00000014”, “CALLERID(num)=0626902626”) in new stack
[Nov 15 11:08:41] VERBOSE[6460][C-00000014] pbx.c: – Executing [01914265847@outgoing:2] Dial(“SIP/9001-00000014”, “IAX2/Poivy/01914265847”) in new stack
[Nov 15 11:08:41] VERBOSE[6460][C-00000014] app_dial.c: – Called IAX2/Poivy/01914265847
[Nov 15 11:08:45] NOTICE[3516] chan_iax2.c: Auto-congesting call due to slow response
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] app_dial.c: – IAX2/Poivy-49 is circuit-busy
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] chan_iax2.c: – Hungup ‘IAX2/Poivy-49’
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] pbx.c: – Auto fallthrough, channel ‘SIP/9001-00000014’ status is ‘CONGESTION’
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] pbx.c: – Executing [h@outgoing:1] Set(“SIP/9001-00000014”, “CALLERID(num)=0626902626”) in new stack
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] pbx.c: – Executing [h@outgoing:2] Dial(“SIP/9001-00000014”, “IAX2/Poivy/h”) in new stack
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] app_dial.c: – Called IAX2/Poivy/h
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] chan_iax2.c: – Hungup ‘IAX2/Poivy-355’
[Nov 15 11:08:45] VERBOSE[6460][C-00000014] pbx.c: == Spawn extension (outgoing, h, 2) exited non-zero on ‘SIP/9001-00000014’
[Nov 15 11:08:51] VERBOSE[3528][C-00000015] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 15 11:08:51] VERBOSE[6461][C-00000015] pbx.c: – Executing [0491557788@outgoing:1] Set(“SIP/9001-00000015”, “CALLERID(num)=0626902626”) in new stack
[Nov 15 11:08:51] VERBOSE[6461][C-00000015] pbx.c: – Executing [0491557788@outgoing:2] Dial(“SIP/9001-00000015”, “IAX2/Poivy/0491557788”) in new stack
[Nov 15 11:08:51] VERBOSE[6461][C-00000015] app_dial.c: – Called IAX2/Poivy/0491557788
[Nov 15 11:08:55] NOTICE[3514] chan_iax2.c: Auto-congesting call due to slow response
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] app_dial.c: – IAX2/Poivy-1226 is circuit-busy
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] chan_iax2.c: – Hungup ‘IAX2/Poivy-1226’
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] pbx.c: – Auto fallthrough, channel ‘SIP/9001-00000015’ status is ‘CONGESTION’
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] pbx.c: – Executing [h@outgoing:1] Set(“SIP/9001-00000015”, “CALLERID(num)=0626902626”) in new stack
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] pbx.c: – Executing [h@outgoing:2] Dial(“SIP/9001-00000015”, “IAX2/Poivy/h”) in new stack
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] app_dial.c: – Called IAX2/Poivy/h
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] chan_iax2.c: – Hungup ‘IAX2/Poivy-5136’
[Nov 15 11:08:55] VERBOSE[6461][C-00000015] pbx.c: == Spawn extension (outgoing, h, 2) exited non-zero on ‘SIP/9001-00000015’
[Nov 15 11:09:04] VERBOSE[6459] asterisk.c: – Remote UNIX connection disconnected[/quote]

david55 · November 15, 2014, 10:33am

You didn’t enable the debug category on the log.

I’m not familiar with IAX2 messages, but I would assume that it is getting no response from the other end. That would typically be a firewall or NAT problem.

gregchelli · November 15, 2014, 12:20pm

i forget that on my post sorry :

debug :

[code][Nov 15 12:53:34] Asterisk 11.14.0 built by root @ gregchelli on a x86_64 running Linux on 2014-11-13 16:28:34 UTC
[Nov 15 12:53:34] DEBUG[5024] config.c: Parsing /etc/asterisk/logger.conf
[Nov 15 12:54:18] DEBUG[3534] chan_sip.c: = Looking for Call ID: YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA. (Checking From) --From tag 79546172 --To-tag
[Nov 15 12:54:18] DEBUG[3534] acl.c: For destination ‘192.168.1.28’, our source address is ‘192.168.1.12’.
[Nov 15 12:54:18] DEBUG[3534] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.12:5060
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: Splitting ‘192.168.1.28:5060’ into…
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: …host ‘192.168.1.28’ and port ‘5060’.
[Nov 15 12:54:18] DEBUG[3534] chan_sip.c: Allocating new SIP dialog for YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA. - INVITE (No RTP)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Begin: parsing SIP “Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri”
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found SIP option: -replaces-
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Matched SIP option: replaces
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found SIP option: -norefersub-
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Matched SIP option: norefersub
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found SIP option: -extended-refer-
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found no match for SIP option: extended-refer (Please file bug report!)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found SIP option: -timer-
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Matched SIP option: timer
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found SIP option: -X-cisco-serviceuri-
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] sip/reqresp_parser.c: Found private SIP option, not supported: X-cisco-serviceuri
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.28:5060’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.28’ and port ‘5060’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.1.28:5060
[Nov 15 12:54:18] DEBUG[3534] chan_sip.c: = Looking for Call ID: YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA. (Checking From) --From tag 79546172 --To-tag as3d493a5d
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Stopping retransmission on ‘YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.’ of Response 1: Match Found
[Nov 15 12:54:18] DEBUG[3534] chan_sip.c: = Looking for Call ID: YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA. (Checking From) --From tag 79546172 --To-tag
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.28:5060’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.28’ and port ‘5060’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] res_rtp_asterisk.c: Allocated port 15902 for RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: RTP instance ‘0x7f9674041b38’ is setup and ready to go
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Setting NAT on RTP to Off
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing session-level SDP o=Zoiper_user 0 0 IN IP4 192.168.1.28… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing session-level SDP s=Zoiper_session… UNSUPPORTED OR FAILED.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.28’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.28’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.28… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 3 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 0 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 8 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 110 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 98 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Setting payload 101 based on m type on 0x7f967b659f30
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 iLBC/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:98 mode=30… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 0 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 3 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 8 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 98 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 101 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] rtp_engine.c: Copying payload 110 from 0x7f967b659f30 to 0x7f9674041d00
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: We’re settling with these formats: (gsm|ulaw|alaw)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Checking SIP call limits for device 900
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Updating call counter for incoming call
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: Splitting ‘192.168.1.12’ into…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] netsock2.c: …host ‘192.168.1.12’ and port ‘’.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Incoming INVITE with ‘timer’ option supported
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] format_pref.c: Could not find preferred codec - Going for the best codec
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: *** Our native formats are (ulaw)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: *** Joint capabilities are (gsm|ulaw|alaw)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: This channel will not be able to handle video.
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: build_route: Contact hop: sip:9001@192.168.1.28:5060;transport=UDP
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: SIP/9001-0000000c: New call is still down… Trying…
[Nov 15 12:54:18] DEBUG[3534][C-0000000c] chan_sip.c: Trying to put ‘SIP/2.0 100’ onto UDP socket destined for 192.168.1.28:5060
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for SIP - 9001
[Nov 15 12:54:18] DEBUG[3271] chan_sip.c: Checking device state for peer 9001
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: Changing state for SIP/9001 - state 1 (Not in use)
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: device ‘SIP/9001’ state ‘1’
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] pbx.c: Launching ‘Set’
[Nov 15 12:54:18] DEBUG[3583] app_queue.c: Device ‘SIP/9001’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] pbx.c: Result of ‘EXTEN’ is ‘0491554477’
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] pbx.c: Launching ‘Dial’
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] chan_iax2.c: ip callno count incremented to 1 for 77.72.169.134
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel_internal_api.c: Channel Call ID changing from [C-0000000c] to [C-0000000c]
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel_internal_api.c: Channel Call ID changing from [C-0000000c] to [C-0000000c]
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] rtp_engine.c: Can’t find native functions for channel ‘IAX2/Poivy-242’
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] chan_iax2.c: prepending gsm to prefs
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel.c: Set channel IAX2/Poivy-242 to read format slin
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel.c: Set channel SIP/9001-0000000c to write format slin
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel.c: Set channel SIP/9001-0000000c to read format slin
[Nov 15 12:54:18] DEBUG[5028][C-0000000c] channel.c: Set channel IAX2/Poivy-242 to write format slin
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for IAX2 - Poivy
[Nov 15 12:54:18] DEBUG[3271] chan_iax2.c: Checking device state for device Poivy
[Nov 15 12:54:18] DEBUG[3271] chan_iax2.c: Found peer. What’s device state of Poivy? addr=1296607622, defaddr=0 maxms=0, lastms=0
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: Changing state for IAX2/Poivy - state 6 (Ringing)
[Nov 15 12:54:18] DEBUG[3271] devicestate.c: device ‘IAX2/Poivy’ state ‘6’
[Nov 15 12:54:18] DEBUG[3583] app_queue.c: Device ‘IAX2/Poivy’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:22] NOTICE[3455] chan_iax2.c: Auto-congesting call due to slow response
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Hanging up channel ‘IAX2/Poivy-242’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_iax2.c: We’re hanging up IAX2/Poivy-242 now…
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for IAX2 - Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Checking device state for device Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Found peer. What’s device state of Poivy? addr=1296607622, defaddr=0 maxms=0, lastms=0
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_sip.c: Trying to put ‘SIP/2.0 503’ onto UDP socket destined for 192.168.1.28:5060
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_sip.c: Setting SIP_ALREADYGONE on dialog YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Soft-Hanging up channel ‘SIP/9001-0000000c’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Soft-Hanging up channel ‘SIP/9001-0000000c’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Soft-Hanging up channel ‘SIP/9001-0000000c’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] pbx.c: Launching ‘Set’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] pbx.c: Result of ‘EXTEN’ is ‘h’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] pbx.c: Launching ‘Dial’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_iax2.c: ip callno count incremented to 2 for 77.72.169.134
[Nov 15 12:54:22] DEBUG[3534] chan_sip.c: = Looking for Call ID: YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA. (Checking From) --From tag 79546172 --To-tag as3f33eb02
[Nov 15 12:54:22] DEBUG[3534][C-0000000c] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 15 12:54:22] DEBUG[3534][C-0000000c] chan_sip.c: Stopping retransmission on ‘YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.’ of Response 2: Match Found
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: Changing state for IAX2/Poivy - state 0 (Unknown)
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: device ‘IAX2/Poivy’ state ‘0’
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for SIP - 9001
[Nov 15 12:54:22] DEBUG[3271] chan_sip.c: Checking device state for peer 9001
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: Changing state for SIP/9001 - state 1 (Not in use)
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: device ‘SIP/9001’ state ‘1’
[Nov 15 12:54:22] DEBUG[3583] app_queue.c: Device ‘IAX2/Poivy’ changed to state ‘0’ (Unknown) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:22] DEBUG[3583] app_queue.c: Device ‘SIP/9001’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel_internal_api.c: Channel Call ID changing from [C-0000000c] to [C-0000000c]
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel_internal_api.c: Channel Call ID changing from [C-0000000c] to [C-0000000c]
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] rtp_engine.c: Can’t find native functions for channel ‘IAX2/Poivy-1326’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_iax2.c: prepending gsm to prefs
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for IAX2 - Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Checking device state for device Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Found peer. What’s device state of Poivy? addr=1296607622, defaddr=0 maxms=0, lastms=0
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: Changing state for IAX2/Poivy - state 6 (Ringing)
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: device ‘IAX2/Poivy’ state ‘6’
[Nov 15 12:54:22] DEBUG[3583] app_queue.c: Device ‘IAX2/Poivy’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Set channel IAX2/Poivy-1326 to read format slin
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Set channel IAX2/Poivy-1326 to write format slin
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Hanging up channel ‘IAX2/Poivy-1326’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_iax2.c: We’re hanging up IAX2/Poivy-1326 now…
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for IAX2 - Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Checking device state for device Poivy
[Nov 15 12:54:22] DEBUG[3271] chan_iax2.c: Found peer. What’s device state of Poivy? addr=1296607622, defaddr=0 maxms=0, lastms=0
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: Changing state for IAX2/Poivy - state 0 (Unknown)
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: device ‘IAX2/Poivy’ state ‘0’
[Nov 15 12:54:22] DEBUG[3583] app_queue.c: Device ‘IAX2/Poivy’ changed to state ‘0’ (Unknown) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] pbx.c: Spawn extension (outgoing,h,2) exited non-zero on ‘SIP/9001-0000000c’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] channel.c: Hanging up channel ‘SIP/9001-0000000c’
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] chan_sip.c: Hangup call SIP/9001-0000000c, SIP callid YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.
[Nov 15 12:54:22] DEBUG[5028][C-0000000c] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: No provider found, checking channel drivers for SIP - 9001
[Nov 15 12:54:22] DEBUG[3271] chan_sip.c: Checking device state for peer 9001
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: Changing state for SIP/9001 - state 1 (Not in use)
[Nov 15 12:54:22] DEBUG[3271] devicestate.c: device ‘SIP/9001’ state ‘1’
[Nov 15 12:54:22] DEBUG[3583] app_queue.c: Device ‘SIP/9001’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 15 12:54:23] DEBUG[3534] chan_sip.c: Session timer stopped: -1 - YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.
[Nov 15 12:54:23] DEBUG[3534] chan_sip.c: Destroying SIP dialog YzA4NDE5NjYxYTg3MDcwYTZkNjlmNmMzNDVlOTM5OTA.
[Nov 15 12:54:23] DEBUG[3534] rtp_engine.c: Destroyed RTP instance ‘0x7f9674041b38’
[Nov 15 12:54:32] DEBUG[3433] chan_iax2.c: Really destroying 242 now…
[Nov 15 12:54:32] DEBUG[3433] chan_iax2.c: schedule decrement of callno used for 77.72.169.134 in 60 seconds
[Nov 15 12:54:32] DEBUG[3433] chan_iax2.c: Really destroying 1326 now…
[Nov 15 12:54:32] DEBUG[3433] chan_iax2.c: schedule decrement of callno used for 77.72.169.134 in 60 seconds

[/code]
mt 5060 and 4569 port is open in the NAT/PAT router’s rules (UDP and TCP)
I turn off the firewall of my computer and the router firewall

Maybe its because VM workstation ? in my router panel i see the VMworkstation linux session is considered as a diffirent computer in the network (my computer is 192.168.1.28 and linux VMworkstation session as 192.168.1.12 (so i use 192.168.1.12 as host for connect to my 9001 sip session in Zoiper free)

And my sip provider account work well , i tested some call with zoiper with my sip poivy session
(sip.poivy.com) and that work well , i sent them a email to check if all work in their side et they replied me its OK

I tested with SFLphone on my linux session too and dont work too

thanks

gregchelli · November 15, 2014, 4:13pm

re , i can call SIP protocol but no caller id , so still the same

So my sip.conf is :

[code][general]
context=greg-poivy
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => gregchelli:tekken552@sip.poivy.com

[210]
type=friend
host=dynamic
username=210
secret=4321
context=appel-sortant
callerid=“210” <0626902626>
nat=yes
language=fr

[forfait-poivy]
type=peer
host=sip.poivy.com
context=poivy-sip
language=fr
username=gregchelli
secret=tekken552
nat=yes
dtmfmode=auto
video=no
[/code]

extensions.conf

[appel-sortant] exten => _x.,1,Set(CALLERID(num)=0626902626) exten => _X.,n,Dial(SIP/${EXTEN}@forfait-poivy)

david55 · November 15, 2014, 6:11pm

Have you cleared the caller ID you intend to present with your ITSP Only dodgy ITSPs accept arbitrary CLI.

gregchelli · November 15, 2014, 6:30pm

Sorry i dont really understand ? U mean say to my ITSP if i can set a callerID ?
no i didnt
and i have resgistered any number with my ITSP
also i tried to put restrictcid=no on sip.conf for both user but still juste show “private num” when i
try to call my cellphone

thanks

seer_m · November 17, 2014, 7:10am

Wait. You’re using IAX to dial from Asterisk to your provider, which doesn’t work. But using your provider details in Zoiper works? Have you tried writing your provider details into Asterisk’s sip.conf as a peer and using SIP instead of IAX to dial?

gregchelli · November 17, 2014, 7:13pm

Yes that work well with SIP ! But i dont have any callerID and callerID is why i use asterisk

thanks for you reply

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