# Phone rings, no voice when answerd

**URL:** <https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584>\
**Category:** Asterisk Support\
**Created:** [November 18, 2006, 6:22pm UTC](https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584 "2006-11-18T18:22:31Z")\
**Posts on this page:** 4\
**Page:** 1

<div class="post-metadata">

**Author:** ![niesta](https://avatars.discourse-cdn.com/v4/letter/n/df705f/32.png) [@niesta](https://community.asterisk.org/u/niesta)\
**Post date:** [November 18, 2006, 6:22pm UTC](https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584/1 "2006-11-18T18:22:31Z")

</div>

Finaly I can make calls, but when a call is answerd there is no voice / sound.

Log files show call answered.

Situation:  
Asterisk server inside Datacenter.  
Client : Xlite (for testing)

Tail of CLI

`Nov 18 16:09:25 NOTICE[23806]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!
Nov 18 16:09:25 WARNING[23806]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3'
Nov 18 16:09:25 WARNING[23806]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player
    -- Executing SetCallerID("SIP/xlite1-2f73", "4153574000") in new stack
    -- Executing Dial("SIP/xlite1-2f73", "IAX2/13252@xxxx/011556281261373") in new stack
    -- Called 13252@voipjet/011556281261373
    -- Call accepted by xx.xxx.xxx.xx (format ulaw)
    -- Format for call is ulaw
    -- IAX2/xxxx/1 is making progress passing it to SIP/xlite1-2f73
    -- IAX2/xxxx/1 is making progress passing it to SIP/xlite1-2f73
    -- IAX2/xxxx/1 answered SIP/xlite1-2f73
Nov 18 16:15:41 WARNING[23806]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call NjQ2ZWUzYjk1ZGM5NDc3OThjYzU1NGQ2NDlmYzBkYmI. for seqno 2 (Non-critical Response)
    -- Hungup 'IAX2/xxxx/1'
  == Spawn extension (default, 011556281261373, 2) exited non-zero on 'SIP/xlite1-2f73'
deusefiel*CLI>`

Who can help me ?

Thanks

---

<div class="post-metadata">

**Author:** ![808blogger](https://avatars.discourse-cdn.com/v4/letter/8/b5e925/32.png) [@808blogger](https://community.asterisk.org/u/808blogger)\
**Post date:** [November 18, 2006, 7:22pm UTC](https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584/2 "2006-11-18T19:22:58Z")

</div>

is the client outside of the LAN?

if it is make sure that in sip.conf you have

nat=yes  
host=dynamic

[quote=“niesta”]Finaly I can make calls, but when a call is answerd there is no voice / sound.

Log files show call answered.

Situation:  
Asterisk server inside Datacenter.  
Client : Xlite (for testing)

Tail of CLI

`Nov 18 16:09:25 NOTICE[23806]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!
Nov 18 16:09:25 WARNING[23806]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3'
Nov 18 16:09:25 WARNING[23806]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player
    -- Executing SetCallerID("SIP/xlite1-2f73", "4153574000") in new stack
    -- Executing Dial("SIP/xlite1-2f73", "IAX2/13252@xxxx/011556281261373") in new stack
    -- Called 13252@voipjet/011556281261373
    -- Call accepted by xx.xxx.xxx.xx (format ulaw)
    -- Format for call is ulaw
    -- IAX2/xxxx/1 is making progress passing it to SIP/xlite1-2f73
    -- IAX2/xxxx/1 is making progress passing it to SIP/xlite1-2f73
    -- IAX2/xxxx/1 answered SIP/xlite1-2f73
Nov 18 16:15:41 WARNING[23806]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call NjQ2ZWUzYjk1ZGM5NDc3OThjYzU1NGQ2NDlmYzBkYmI. for seqno 2 (Non-critical Response)
    -- Hungup 'IAX2/xxxx/1'
  == Spawn extension (default, 011556281261373, 2) exited non-zero on 'SIP/xlite1-2f73'
deusefiel*CLI>`

Who can help me ?

Thanks[/quote]

---

<div class="post-metadata">

**Author:** ![niesta](https://avatars.discourse-cdn.com/v4/letter/n/df705f/32.png) [@niesta](https://community.asterisk.org/u/niesta)\
**Post date:** [November 19, 2006, 12:28pm UTC](https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584/3 "2006-11-19T12:28:55Z")

</div>

Thanks for your answer.

Yes, the client is out of the NAT, but still no go.

phone rings, still no sound / voice.

Asterisk is running on Debian in a DC (USA)  
Client is X-lite on windows (Brazil)

Here is CLI reload output and afterwards a call.

# [code]root@deusefiel:~# asterisk -r Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k, Copyright © 1999-2004 Digium. Written by Mark Spencer [markster@digium.com](mailto:markster@digium.com)

Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k currently running on deusefiel (pid = 23805)  
nectionel_CLI\>  
Verbosity is at least 9  
deusefiel_CLI\> reload  
== Parsing ‘/etc/asterisk/extconfig.conf’: Found  
== Parsing ‘/etc/asterisk/manager.conf’: Found  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
== Parsing ‘/etc/asterisk/rtp.conf’: Found  
== RTP Allocating from port range 14010 -\> 14030  
– Reloading module ‘res\_crypto.so’ (Cryptographic Digital Signatures)  
– Reloading module ‘res\_adsi.so’ (ADSI Resource)  
== Parsing ‘/etc/asterisk/adsi.conf’: Found  
– Reloading module ‘res\_indications.so’ (Indications Configuration)  
– Unregistered indication country ‘cl’  
– Unregistered indication country 'tw’  
Nov 19 10:03:20 NOTICE[23806]: indications.c:397 ast\_unregister\_indication\_country: Removed default indication country ‘us’  
– Unregistered indication country ‘us’  
– Unregistered indication country ‘au’  
– Unregistered indication country ‘fr’  
– Unregistered indication country ‘de’  
– Unregistered indication country ‘nl’  
– Unregistered indication country ‘uk’  
– Unregistered indication country ‘fi’  
– Unregistered indication country ‘no’  
– Unregistered indication country ‘br’  
– Unregistered indication country ‘za’  
– Unregistered indication country ‘it’  
– Unregistered indication country ‘us-o’  
– Unregistered indication country ‘gr’  
– Unregistered indication country ‘ru’  
– Unregistered indication country ‘nz’  
== Parsing ‘/etc/asterisk/indications.conf’: Found  
– Registered indication country ‘cl’  
– Registered indication country ‘tw’  
– Registered indication country ‘us’  
– Registered indication country ‘au’  
– Registered indication country ‘fr’  
– Registered indication country ‘de’  
– Registered indication country ‘nl’  
– Registered indication country ‘uk’  
– Registered indication country ‘fi’  
– Registered indication country ‘no’  
– Registered indication country ‘br’  
– Registered indication country ‘za’  
– Registered indication country ‘it’  
– Registered indication country ‘us-o’  
– Registered indication country ‘gr’  
– Registered indication country ‘ru’  
– Registered indication country ‘nz’  
– Setting default indication country to ‘us’  
– Reloading module ‘app\_queue.so’ (True Call Queueing)  
== Parsing ‘/etc/asterisk/queues.conf’: Found  
– Reloading module ‘cdr\_csv.so’ (Comma Separated Values CDR Backend)  
– Reloading module ‘cdr\_sqlite.so’ (SQLite CDR Backend)  
– Reloading module ‘cdr\_odbc.so’ (ODBC CDR Backend)  
== Unregistered ‘ODBC’ CDR backend  
== Parsing ‘/etc/asterisk/cdr\_odbc.conf’: Found  
– Reloading module ‘app\_enumlookup.so’ (ENUM Lookup)  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
– Reloading module ‘cdr\_pgsql.so’ (PostgreSQL CDR Backend)  
== Unregistered ‘pgsql’ CDR backend  
== Parsing ‘/etc/asterisk/cdr\_pgsql.conf’: Found  
– Reloading module ‘chan\_skinny.so’ (Skinny Client Control Protocol (Skinny))  
– Reloading module ‘app\_voicemail.so’ (Comedian Mail (Voicemail System))  
== Parsing ‘/etc/asterisk/voicemail.conf’: Found  
– Reloading module ‘app\_txtcidname.so’ (TXTCIDName)  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
– Reloading module ‘pbx\_config.so’ (Text Extension Configuration)  
== Parsing ‘/etc/asterisk/extensions.conf’: Found  
– Setting global variable ‘CONSOLE’ to ‘Console/dsp’  
– Setting global variable ‘IAXINFO’ to ‘guest’  
– Setting global variable ‘TRUNK’ to ‘Zap/g2’  
– Setting global variable ‘TRUNKMSD’ to ‘1’  
– Registered extension context ‘iaxtel700’  
– Added extension ‘\_91700XXXXXXX’ priority 1 to iaxtel700  
– Registered extension context ‘iaxprovider’  
– Registered extension context ‘trunkint’  
– Added extension ‘\_9011.’ priority 1 to trunkint  
– Added extension ‘\_9011.’ priority 2 to trunkint  
– Registered extension context ‘trunkld’  
– Added extension ‘\_91NXXNXXXXXX’ priority 1 to trunkld  
– Added extension ‘\_91NXXNXXXXXX’ priority 2 to trunkld  
– Registered extension context ‘trunklocal’  
– Added extension ‘\_9NXXXXXX’ priority 1 to trunklocal  
– Added extension ‘\_9NXXXXXX’ priority 2 to trunklocal  
– Registered extension context ‘trunktollfree’  
– Added extension ‘\_91800NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91800NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91888NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91888NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91877NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91877NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91866NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91866NXXXXXX’ priority 2 to trunktollfree  
– Registered extension context ‘international’  
– Including context ‘longdistance’ in context ‘international’  
– Including context ‘trunkint’ in context ‘international’  
– Registered extension context ‘longdistance’  
– Including context ‘local’ in context ‘longdistance’  
– Including context ‘trunkld’ in context ‘longdistance’  
– Registered extension context ‘local’  
– Including context ‘default’ in context ‘local’  
– Including context ‘parkedcalls’ in context ‘local’  
– Including context ‘trunklocal’ in context ‘local’  
– Including context ‘iaxtel700’ in context ‘local’  
– Including context ‘trunktollfree’ in context ‘local’  
– Including context ‘iaxprovider’ in context ‘local’  
– Registered extension context ‘macro-stdexten’  
– Added extension ‘s’ priority 1 to macro-stdexten  
– Added extension ‘s’ priority 2 to macro-stdexten  
– Added extension ‘s-NOANSWER’ priority 1 to macro-stdexten  
– Added extension ‘s-NOANSWER’ priority 2 to macro-stdexten  
– Added extension ‘s-BUSY’ priority 1 to macro-stdexten  
– Added extension ‘s-BUSY’ priority 2 to macro-stdexten  
– Added extension ‘\_s-.’ priority 1 to macro-stdexten  
– Added extension ‘a’ priority 1 to macro-stdexten  
– Registered extension context ‘demo’  
– Added extension ‘s’ priority 1 to demo  
– Added extension ‘s’ priority 2 to demo  
– Added extension ‘s’ priority 3 to demo  
– Added extension ‘s’ priority 4 to demo  
– Added extension ‘s’ priority 5 to demo  
– Added extension ‘s’ priority 6 to demo  
– Added extension ‘2’ priority 1 to demo  
– Added extension ‘2’ priority 2 to demo  
– Added extension ‘3’ priority 1 to demo  
– Added extension ‘3’ priority 2 to demo  
– Added extension ‘1000’ priority 1 to demo  
– Added extension ‘1234’ priority 1 to demo  
– Added extension ‘1234’ priority 2 to demo  
– Added extension ‘1235’ priority 1 to demo  
– Added extension ‘1236’ priority 1 to demo  
– Added extension ‘1236’ priority 2 to demo  
– Added extension ‘#’ priority 1 to demo  
– Added extension ‘#’ priority 2 to demo  
– Added extension ‘t’ priority 1 to demo  
– Added extension ‘i’ priority 1 to demo  
– Added extension ‘500’ priority 1 to demo  
– Added extension ‘500’ priority 2 to demo  
– Added extension ‘500’ priority 3 to demo  
– Added extension ‘500’ priority 4 to demo  
– Added extension ‘600’ priority 1 to demo  
– Added extension ‘600’ priority 2 to demo  
– Added extension ‘600’ priority 3 to demo  
– Added extension ‘600’ priority 4 to demo  
– Added extension ‘8500’ priority 1 to demo  
– Added extension ‘8500’ priority 2 to demo  
– Registered extension context ‘default’  
– Including context ‘demo’ in context ‘default’  
– Added extension ‘\_1NXXNXXXXXX’ priority 1 to default  
– Added extension ‘\_1NXXNXXXXXX’ priority 2 to default  
– Added extension ‘\_011.’ priority 1 to default  
– Added extension ‘\_011.’ priority 2 to default  
– Reloading module ‘chan\_iax2.so’ (Inter Asterisk eXchange (Ver 2))  
== Parsing ‘/etc/asterisk/iax.conf’: Found  
== Loaded firmware ‘iaxy.bin’  
== Parsing ‘/etc/asterisk/iaxprov.conf’: Found  
– Loaded provisioning template ‘default’  
– Reloading module ‘chan\_sip.so’ (Session Initiation Protocol (SIP))  
– Reloading module ‘chan\_mgcp.so’ (Media Gateway Control Protocol (MGCP))  
Reloading SIP  
Reloading MGCP  
== Parsing ‘/etc/asterisk/mgcp.conf’: Found  
– Reloading module ‘chan\_agent.so’ (Agent Proxy Channel)  
== Parsing ‘/etc/asterisk/agents.conf’: Found  
== Parsing ‘/etc/asterisk/sip.conf’: Found  
== MGCP Listening on 0.0.0.0:2727  
== Using TOS bits 0  
– Reloading module ‘cdr\_manager.so’ (Asterisk Call Manager CDR Backend)  
== Parsing ‘/etc/asterisk/cdr\_manager.conf’: Found  
– Reloading module ‘chan\_local.so’ (Local Proxy Channel)  
– Executing SetCallerID(“SIP/xlite1-25c6”, “4153574000”) in new stack  
– Executing Dial(“SIP/xlite1-25c6”, “IAX2/xxxx@xxxx/011556235980627”) in new stack  
– Called xxxx@xxxx/011556235980627  
– Call accepted by 8.11.164.234 (format ulaw)  
– Format for call is ulaw  
– IAX2/voipjet/1 is making progress passing it to SIP/xlite1-25c6  
Nov 19 10:04:29 NOTICE[23806]: res\_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!  
Nov 19 10:04:29 WARNING[23806]: res\_musiconhold.c:205 spawn\_mp3: Found no files in '/usr/share/asterisk/mohmp3’  
Nov 19 10:04:29 WARNING[23806]: res\_musiconhold.c:278 monmp3thread: unable to spawn mp3player  
– Hungup ‘IAX2/xxxx/1’  
== Spawn extension (default, 011556235980627, 2) exited non-zero on ‘SIP/xlite1-25c6’  
– Executing SetCallerID(“SIP/xlite1-44d7”, “4153574000”) in new stack  
– Executing Dial(“SIP/xlite1-44d7”, “IAX2/xxxx@xxxx/011556281261373”) in new stack  
– Called xxxx@xxxx/011556281261373  
– Call accepted by [xx.xxx.xxx.xxx](http://xx.xxx.xxx.xxx) (format ulaw)  
– Format for call is ulaw  
– IAX2/xxxx/2 is making progress passing it to SIP/xlite1-44d7  
– IAX2/xxxx/2 is making progress passing it to SIP/xlite1-44d7  
– IAX2/xxxx/2 answered SIP/xlite1-44d7  
Nov 19 10:05:31 WARNING[23806]: chan\_sip.c:694 retrans\_pkt: Maximum retries exceeded on call MDM4NjdlNjRhMWRkZWUyNzgzMmQxOTAzNmQ1YWZhMTg. for seqno 2 (Non-critical Response)  
– Hungup ‘IAX2/xxxx/2’  
== Spawn extension (default, 011556281261373, 2) exited non-zero on 'SIP/xlite1-44d7’  
deusefiel\*CLI\> quit  
root@deusefiel:~#[/code]

---

<div class="post-metadata">

**Author:** ![808blogger](https://avatars.discourse-cdn.com/v4/letter/8/b5e925/32.png) [@808blogger](https://community.asterisk.org/u/808blogger)\
**Post date:** [November 19, 2006, 7:13pm UTC](https://community.asterisk.org/t/phone-rings-no-voice-when-answerd/10584/4 "2006-11-19T19:13:20Z")

</div>

what does sip debug output say?

is this setup working like this?

xlite/sip —\>NAT-----\>INTERNET--------\>NAT ------\>asterisk-----\>NAT-----\>TELCOPROVIDER/IAX2

take a look here.  
[voip-info.org/wiki-Asterisk+ … +solutions](http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions)

I think you fall under condition #4  
Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk

[fridu.org/index.php?option=c … &Itemid=55](http://www.fridu.org/index.php?option=com_content&task=category&sectionid=6&id=27&Itemid=55)

it may be MUCH easier if you can vpn the 2 offices together and run the sip over the vpn. otherwise try using idefsk (asteriskguru) softphone as it is IAX and may work better over the multi nat scenario.

[quote=“niesta”]Thanks for your answer.

Yes, the client is out of the NAT, but still no go.

phone rings, still no sound / voice.

Asterisk is running on Debian in a DC (USA)  
Client is X-lite on windows (Brazil)

Here is CLI reload output and afterwards a call.

# [code]root@deusefiel:~# asterisk -r Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k, Copyright © 1999-2004 Digium. Written by Mark Spencer [markster@digium.com](mailto:markster@digium.com)

Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k currently running on deusefiel (pid = 23805)  
nectionel_CLI\>  
Verbosity is at least 9  
deusefiel_CLI\> reload  
== Parsing ‘/etc/asterisk/extconfig.conf’: Found  
== Parsing ‘/etc/asterisk/manager.conf’: Found  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
== Parsing ‘/etc/asterisk/rtp.conf’: Found  
== RTP Allocating from port range 14010 -\> 14030  
– Reloading module ‘res\_crypto.so’ (Cryptographic Digital Signatures)  
– Reloading module ‘res\_adsi.so’ (ADSI Resource)  
== Parsing ‘/etc/asterisk/adsi.conf’: Found  
– Reloading module ‘res\_indications.so’ (Indications Configuration)  
– Unregistered indication country ‘cl’  
– Unregistered indication country 'tw’  
Nov 19 10:03:20 NOTICE[23806]: indications.c:397 ast\_unregister\_indication\_country: Removed default indication country ‘us’  
– Unregistered indication country ‘us’  
– Unregistered indication country ‘au’  
– Unregistered indication country ‘fr’  
– Unregistered indication country ‘de’  
– Unregistered indication country ‘nl’  
– Unregistered indication country ‘uk’  
– Unregistered indication country ‘fi’  
– Unregistered indication country ‘no’  
– Unregistered indication country ‘br’  
– Unregistered indication country ‘za’  
– Unregistered indication country ‘it’  
– Unregistered indication country ‘us-o’  
– Unregistered indication country ‘gr’  
– Unregistered indication country ‘ru’  
– Unregistered indication country ‘nz’  
== Parsing ‘/etc/asterisk/indications.conf’: Found  
– Registered indication country ‘cl’  
– Registered indication country ‘tw’  
– Registered indication country ‘us’  
– Registered indication country ‘au’  
– Registered indication country ‘fr’  
– Registered indication country ‘de’  
– Registered indication country ‘nl’  
– Registered indication country ‘uk’  
– Registered indication country ‘fi’  
– Registered indication country ‘no’  
– Registered indication country ‘br’  
– Registered indication country ‘za’  
– Registered indication country ‘it’  
– Registered indication country ‘us-o’  
– Registered indication country ‘gr’  
– Registered indication country ‘ru’  
– Registered indication country ‘nz’  
– Setting default indication country to ‘us’  
– Reloading module ‘app\_queue.so’ (True Call Queueing)  
== Parsing ‘/etc/asterisk/queues.conf’: Found  
– Reloading module ‘cdr\_csv.so’ (Comma Separated Values CDR Backend)  
– Reloading module ‘cdr\_sqlite.so’ (SQLite CDR Backend)  
– Reloading module ‘cdr\_odbc.so’ (ODBC CDR Backend)  
== Unregistered ‘ODBC’ CDR backend  
== Parsing ‘/etc/asterisk/cdr\_odbc.conf’: Found  
– Reloading module ‘app\_enumlookup.so’ (ENUM Lookup)  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
– Reloading module ‘cdr\_pgsql.so’ (PostgreSQL CDR Backend)  
== Unregistered ‘pgsql’ CDR backend  
== Parsing ‘/etc/asterisk/cdr\_pgsql.conf’: Found  
– Reloading module ‘chan\_skinny.so’ (Skinny Client Control Protocol (Skinny))  
– Reloading module ‘app\_voicemail.so’ (Comedian Mail (Voicemail System))  
== Parsing ‘/etc/asterisk/voicemail.conf’: Found  
– Reloading module ‘app\_txtcidname.so’ (TXTCIDName)  
== Parsing ‘/etc/asterisk/enum.conf’: Found  
– Reloading module ‘pbx\_config.so’ (Text Extension Configuration)  
== Parsing ‘/etc/asterisk/extensions.conf’: Found  
– Setting global variable ‘CONSOLE’ to ‘Console/dsp’  
– Setting global variable ‘IAXINFO’ to ‘guest’  
– Setting global variable ‘TRUNK’ to ‘Zap/g2’  
– Setting global variable ‘TRUNKMSD’ to ‘1’  
– Registered extension context ‘iaxtel700’  
– Added extension ‘\_91700XXXXXXX’ priority 1 to iaxtel700  
– Registered extension context ‘iaxprovider’  
– Registered extension context ‘trunkint’  
– Added extension ‘\_9011.’ priority 1 to trunkint  
– Added extension ‘\_9011.’ priority 2 to trunkint  
– Registered extension context ‘trunkld’  
– Added extension ‘\_91NXXNXXXXXX’ priority 1 to trunkld  
– Added extension ‘\_91NXXNXXXXXX’ priority 2 to trunkld  
– Registered extension context ‘trunklocal’  
– Added extension ‘\_9NXXXXXX’ priority 1 to trunklocal  
– Added extension ‘\_9NXXXXXX’ priority 2 to trunklocal  
– Registered extension context ‘trunktollfree’  
– Added extension ‘\_91800NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91800NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91888NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91888NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91877NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91877NXXXXXX’ priority 2 to trunktollfree  
– Added extension ‘\_91866NXXXXXX’ priority 1 to trunktollfree  
– Added extension ‘\_91866NXXXXXX’ priority 2 to trunktollfree  
– Registered extension context ‘international’  
– Including context ‘longdistance’ in context ‘international’  
– Including context ‘trunkint’ in context ‘international’  
– Registered extension context ‘longdistance’  
– Including context ‘local’ in context ‘longdistance’  
– Including context ‘trunkld’ in context ‘longdistance’  
– Registered extension context ‘local’  
– Including context ‘default’ in context ‘local’  
– Including context ‘parkedcalls’ in context ‘local’  
– Including context ‘trunklocal’ in context ‘local’  
– Including context ‘iaxtel700’ in context ‘local’  
– Including context ‘trunktollfree’ in context ‘local’  
– Including context ‘iaxprovider’ in context ‘local’  
– Registered extension context ‘macro-stdexten’  
– Added extension ‘s’ priority 1 to macro-stdexten  
– Added extension ‘s’ priority 2 to macro-stdexten  
– Added extension ‘s-NOANSWER’ priority 1 to macro-stdexten  
– Added extension ‘s-NOANSWER’ priority 2 to macro-stdexten  
– Added extension ‘s-BUSY’ priority 1 to macro-stdexten  
– Added extension ‘s-BUSY’ priority 2 to macro-stdexten  
– Added extension ‘\_s-.’ priority 1 to macro-stdexten  
– Added extension ‘a’ priority 1 to macro-stdexten  
– Registered extension context ‘demo’  
– Added extension ‘s’ priority 1 to demo  
– Added extension ‘s’ priority 2 to demo  
– Added extension ‘s’ priority 3 to demo  
– Added extension ‘s’ priority 4 to demo  
– Added extension ‘s’ priority 5 to demo  
– Added extension ‘s’ priority 6 to demo  
– Added extension ‘2’ priority 1 to demo  
– Added extension ‘2’ priority 2 to demo  
– Added extension ‘3’ priority 1 to demo  
– Added extension ‘3’ priority 2 to demo  
– Added extension ‘1000’ priority 1 to demo  
– Added extension ‘1234’ priority 1 to demo  
– Added extension ‘1234’ priority 2 to demo  
– Added extension ‘1235’ priority 1 to demo  
– Added extension ‘1236’ priority 1 to demo  
– Added extension ‘1236’ priority 2 to demo  
– Added extension ‘#’ priority 1 to demo  
– Added extension ‘#’ priority 2 to demo  
– Added extension ‘t’ priority 1 to demo  
– Added extension ‘i’ priority 1 to demo  
– Added extension ‘500’ priority 1 to demo  
– Added extension ‘500’ priority 2 to demo  
– Added extension ‘500’ priority 3 to demo  
– Added extension ‘500’ priority 4 to demo  
– Added extension ‘600’ priority 1 to demo  
– Added extension ‘600’ priority 2 to demo  
– Added extension ‘600’ priority 3 to demo  
– Added extension ‘600’ priority 4 to demo  
– Added extension ‘8500’ priority 1 to demo  
– Added extension ‘8500’ priority 2 to demo  
– Registered extension context ‘default’  
– Including context ‘demo’ in context ‘default’  
– Added extension ‘\_1NXXNXXXXXX’ priority 1 to default  
– Added extension ‘\_1NXXNXXXXXX’ priority 2 to default  
– Added extension ‘\_011.’ priority 1 to default  
– Added extension ‘\_011.’ priority 2 to default  
– Reloading module ‘chan\_iax2.so’ (Inter Asterisk eXchange (Ver 2))  
== Parsing ‘/etc/asterisk/iax.conf’: Found  
== Loaded firmware ‘iaxy.bin’  
== Parsing ‘/etc/asterisk/iaxprov.conf’: Found  
– Loaded provisioning template ‘default’  
– Reloading module ‘chan\_sip.so’ (Session Initiation Protocol (SIP))  
– Reloading module ‘chan\_mgcp.so’ (Media Gateway Control Protocol (MGCP))  
Reloading SIP  
Reloading MGCP  
== Parsing ‘/etc/asterisk/mgcp.conf’: Found  
– Reloading module ‘chan\_agent.so’ (Agent Proxy Channel)  
== Parsing ‘/etc/asterisk/agents.conf’: Found  
== Parsing ‘/etc/asterisk/sip.conf’: Found  
== MGCP Listening on 0.0.0.0:2727  
== Using TOS bits 0  
– Reloading module ‘cdr\_manager.so’ (Asterisk Call Manager CDR Backend)  
== Parsing ‘/etc/asterisk/cdr\_manager.conf’: Found  
– Reloading module ‘chan\_local.so’ (Local Proxy Channel)  
– Executing SetCallerID(“SIP/xlite1-25c6”, “4153574000”) in new stack  
– Executing Dial(“SIP/xlite1-25c6”, “IAX2/xxxx@xxxx/011556235980627”) in new stack  
– Called xxxx@xxxx/011556235980627  
– Call accepted by 8.11.164.234 (format ulaw)  
– Format for call is ulaw  
– IAX2/voipjet/1 is making progress passing it to SIP/xlite1-25c6  
Nov 19 10:04:29 NOTICE[23806]: res\_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!  
Nov 19 10:04:29 WARNING[23806]: res\_musiconhold.c:205 spawn\_mp3: Found no files in '/usr/share/asterisk/mohmp3’  
Nov 19 10:04:29 WARNING[23806]: res\_musiconhold.c:278 monmp3thread: unable to spawn mp3player  
– Hungup ‘IAX2/xxxx/1’  
== Spawn extension (default, 011556235980627, 2) exited non-zero on ‘SIP/xlite1-25c6’  
– Executing SetCallerID(“SIP/xlite1-44d7”, “4153574000”) in new stack  
– Executing Dial(“SIP/xlite1-44d7”, “IAX2/xxxx@xxxx/011556281261373”) in new stack  
– Called xxxx@xxxx/011556281261373  
– Call accepted by [xx.xxx.xxx.xxx](http://xx.xxx.xxx.xxx) (format ulaw)  
– Format for call is ulaw  
– IAX2/xxxx/2 is making progress passing it to SIP/xlite1-44d7  
– IAX2/xxxx/2 is making progress passing it to SIP/xlite1-44d7  
– IAX2/xxxx/2 answered SIP/xlite1-44d7  
Nov 19 10:05:31 WARNING[23806]: chan\_sip.c:694 retrans\_pkt: Maximum retries exceeded on call MDM4NjdlNjRhMWRkZWUyNzgzMmQxOTAzNmQ1YWZhMTg. for seqno 2 (Non-critical Response)  
– Hungup ‘IAX2/xxxx/2’  
== Spawn extension (default, 011556281261373, 2) exited non-zero on 'SIP/xlite1-44d7’  
deusefiel\*CLI\> quit  
root@deusefiel:~#[/code][/quote]