[Freeswitch-users] Cause: MANDATORY_IE_MISSING - No dial-string available

24 Sep 2008

      Hello,

I have setup Freeswitch with xml_curl and provide configs almost 
identical to the local xml files. I can call external gateways, however 
when I call a local phone, I cannot connect and it drops me to the VM of 
the phone.

2008-09-24 23:30:55 [DEBUG] switch_core_state_machine.c:140 
switch_core_standard_on_execute() sofia/internal/1002@siplocal.my.domain 
Execute db(insert/last_dial_ext/${dialed_ext}/${uuid})
2008-09-24 23:30:55 [DEBUG] switch_core_session.c:1053 
switch_core_session_execute_application() 
sofia/internal/1002@siplocal.my.domain Expanded String 
db(insert/last_dial_ext/1003/1078c6e4-8a80-11dd-8aeb-2f433e89ec8c)
2008-09-24 23:30:55 [DEBUG] switch_core_state_machine.c:140 
switch_core_standard_on_execute() sofia/internal/1002@siplocal.my.domain 
Execute bridge(user/${dialed_ext}@siplocal.my.domain)
2008-09-24 23:30:55 [DEBUG] switch_core_session.c:1053 
switch_core_session_execute_application() 
sofia/internal/1002@siplocal.my.domain Expanded String 
*bridge(user/1003@siplocal.my.domain)*
2008-09-24 23:30:56 [ERR] mod_dptools.c:1957 user_outgoing_channel() *No 
dial-string available, please check your user directory.*
2008-09-24 23:30:56 [ERR] switch_ivr_originate.c:926 
switch_ivr_originate() Cannot create outgoing channel of type [user] 
cause: *[MANDATORY_IE_MISSING]*
2008-09-24 23:30:56 [DEBUG] switch_ivr_originate.c:1404 
switch_ivr_originate() Originate Resulted in *Error Cause: 96 
[MANDATORY_IE_MISSING]*
2008-09-24 23:30:56 [INFO] mod_dptools.c:1794 audio_bridge_function() 
Originate Failed.  Cause: MANDATORY_IE_MISSING

Boths phones 1002 and 1003 are successfully registered in the internal 
context. I can call external numbers and internal numbers like VMs, 
conferences etc. When I use the local xml config files it works. The 
only thing I see is different, is that the external dialplan is a subset 
of the original dialplan.

What can be the cause for this error message? The bridge string seems ok 
for me and is the same as when I call with local config files.

When I regard the log I see the following: So this should be fine.
  <app_log>
    <application app_name="set" app_data="use_profile=nat"></application>
    <application app_name="set_user" 
app_data="default@siplocal.my.domain"></application>
    <application app_name="set" 
app_data="sip_secure_media=true"></application>
    <application app_name="db" 
app_data="insert/spymap/1002/edaeda08-8a85-11dd-a728-e3e019d7122d"></application>
    <application app_name="db" 
app_data="insert/last_dial/1002/1003"></application>
    <application app_name="db" 
app_data="insert/last_dial/global/edaeda08-8a85-11dd-a728-e3e019d7122d"></application>
    <application app_name="set" app_data="dialed_ext=1003"></application>
    <application app_name="export" app_data="dialed_ext=1003"></application>
    <application app_name="bind_meta_app" app_data="1 b s 
execute_extension::dx XML features"></application>
    <application app_name="bind_meta_app" app_data="2 b s 
record_session::/usr/local/freeswitch/recordings/1002.2008-09-25-00-12-51.wav"></application>
    <application app_name="bind_meta_app" app_data="3 b s 
execute_extension::cf XML features"></application>
    <application app_name="set" app_data="transfer_ringback=%(2000, 
4000, 440.0, 480.0)"></application>
    <application app_name="set" app_data="call_timeout=30"></application>
    <application app_name="set" 
app_data="hangup_after_bridge=true"></application>
    <application app_name="set" 
app_data="continue_on_fail=true"></application>
    <application app_name="db" 
app_data="insert/call_return/1003/1002"></application>
    <application app_name="db" 
app_data="insert/last_dial_ext/1003/edaeda08-8a85-11dd-a728-e3e019d7122d"></application>
    <application app_name="bridge" 
app_data="user/1003@siplocal.my.domain"></application>
  </app_log>
  <callflow dialplan="XML" profile_index="1">
    <extension name="global" number="1003" current_app="answer">
      <application app_name="set" 
app_data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? 
nat : default)}"></application>
      <application app_name="set_user" 
app_data="default@${domain}"></application>
      <application app_name="set" 
app_data="sip_secure_media=true"></application>
      <application app_name="db" 
app_data="insert/spymap/${caller_id_number}/${uuid}"></application>
      <application app_name="db" 
app_data="insert/last_dial/${caller_id_number}/${destination_number}"></application>
      <application app_name="db" 
app_data="insert/last_dial/global/${uuid}"></application>
      <application app_name="set" app_data="dialed_ext=1003"></application>
      <application app_name="export" 
app_data="dialed_ext=1003"></application>
      <application app_name="bind_meta_app" app_data="1 b s 
execute_extension::dx XML features"></application>
      <application app_name="bind_meta_app" app_data="2 b s 
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"></application>
      <application app_name="bind_meta_app" app_data="3 b s 
execute_extension::cf XML features"></application>
      <application app_name="set" 
app_data="transfer_ringback=${us-ring}"></application>
      <application app_name="set" app_data="call_timeout=30"></application>
      <application app_name="set" 
app_data="hangup_after_bridge=true"></application>
      <application app_name="set" 
app_data="continue_on_fail=true"></application>
      <application app_name="db" 
app_data="insert/call_return/${dialed_ext}/${caller_id_number}"></application>
      <application app_name="db" 
app_data="insert/last_dial_ext/${dialed_ext}/${uuid}"></application>
      <application app_name="bridge" 
app_data="user/${dialed_ext}@siplocal.my.domain"></application>
      <application last_executed="true" app_name="answer" 
app_data=""></application>
      <application app_name="sleep" app_data="1000"></application>
      <application app_name="voicemail" app_data="default 
siplocal.my.domain ${dialed_ext}"></application>
    </extension>
    <caller_profile>
      <username>1002</username>
      <dialplan>XML</dialplan>
      <caller_id_name>Freeswitch1002</caller_id_name>
      <ani></ani>
      <aniii></aniii>
      <caller_id_number>1002</caller_id_number>
      <network_addr>192.168.178.1</network_addr>
      <rdnis></rdnis>
      <destination_number>1003</destination_number>
      <uuid>edaeda08-8a85-11dd-a728-e3e019d7122d</uuid>
      <source>mod_sofia</source>
      <context>default</context>
      <chan_name>sofia/internal/1002@siplocal.my.domain</chan_name>
      <originatee>
        <originatee_caller_profile>
          <username>1002</username>
          <dialplan>XML</dialplan>
          <caller_id_name>Extension 1002</caller_id_name>
          <ani></ani>
          <aniii></aniii>
          <caller_id_number>1002</caller_id_number>
          <network_addr>192.168.178.1</network_addr>
          <rdnis></rdnis>

<destination_number>1003@192.168.178.1:2127;transport=tls;fs_nat=yes</destination_number>
          <uuid>ee27aaa0-8a85-11dd-a728-e3e019d7122d</uuid>
          <source>mod_sofia</source>
          <context>default</context>

<chan_name>sofia/internal/1003@192.168.178.1:2127;transport=tls;fs_nat=yes</chan_name>
        </originatee_caller_profile>
        <originatee_caller_profile>
          <username>1002</username>
          <dialplan>XML</dialplan>
          <caller_id_name>Extension 1002</caller_id_name>
          <ani></ani>
          <aniii></aniii>
          <caller_id_number>1002</caller_id_number>
          <network_addr>192.168.178.1</network_addr>
          <rdnis></rdnis>

<destination_number>1003@192.168.178.1:2127;transport=tls;fs_nat=yes</destination_number>
          <uuid>ee27aaa0-8a85-11dd-a728-e3e019d7122d</uuid>
          <source>mod_sofia</source>
          <context>default</context>

<chan_name>sofia/internal/1003@192.168.178.1:2127;transport=tls;fs_nat=yes</chan_name>
        </originatee_caller_profile>
      </originatee>
    </caller_profile>
    <times>
      <created_time>1222294371386160</created_time>
      <profile_created_time>1222294371386160</profile_created_time>
      <progress_time>1222294372279689</progress_time>
      <progress_media_time>0</progress_media_time>
      <answered_time>1222294372411530</answered_time>
      <hangup_time>1222294374190141</hangup_time>
      <transfer_time>0</transfer_time>
    </times>
  </callflow>
  <app_log>
    <application app_name="set" app_data="dialed_ext=1003"></application>
    <application app_name="export" app_data="dialed_ext=1003"></application>
    <application app_name="bind_meta_app" app_data="1 b s 
execute_extension::dx XML features"></application>
    <application app_name="bind_meta_app" app_data="2 b s 
record_session::/usr/local/freeswitch/recordings/1002.2008-09-25-00-16-22.wav"></application>
    <application app_name="bind_meta_app" app_data="3 b s 
execute_extension::cf XML features"></application>
    <application app_name="set" app_data="transfer_ringback=%(2000, 
4000, 440.0, 480.0)"></application>
    <application app_name="set" app_data="call_timeout=30"></application>
    <application app_name="set" 
app_data="hangup_after_bridge=true"></application>
    <application app_name="set" 
app_data="continue_on_fail=true"></application>
    <application app_name="db" 
app_data="insert/call_return/1003/1002"></application>
    <application app_name="db" 
app_data="insert/last_dial_ext/1003/69a1337c-8a86-11dd-97c6-c70d0d682f22"></application>
    <application app_name="bridge" 
app_data="user/1003@siplocal.my.domain"></application>
    <application app_name="answer" app_data=""></application>
    <application app_name="sleep" app_data="1000"></application>
    <application app_name="voicemail" app_data="default 
siplocal.my.domain 1003"></application>
  </app_log>
  <callflow dialplan="XML" profile_index="1">
    <extension name="Local_Extension" number="1003">
      <application app_name="set" app_data="dialed_ext=1003"></application>
      <application app_name="export" 
app_data="dialed_ext=1003"></application>
      <application app_name="bind_meta_app" app_data="1 b s 
execute_extension::dx XML features"></application>
      <application app_name="bind_meta_app" app_data="2 b s 
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"></application>
      <application app_name="bind_meta_app" app_data="3 b s 
execute_extension::cf XML features"></application>
      <application app_name="set" 
app_data="transfer_ringback=${us-ring}"></application>
      <application app_name="set" app_data="call_timeout=30"></application>
      <application app_name="set" 
app_data="hangup_after_bridge=true"></application>
      <application app_name="set" 
app_data="continue_on_fail=true"></application>
      <application app_name="db" 
app_data="insert/call_return/${dialed_ext}/${caller_id_number}"></application>
      <application app_name="db" 
app_data="insert/last_dial_ext/${dialed_ext}/${uuid}"></application>
      <application app_name="bridge" 
app_data="user/${dialed_ext}@siplocal.my.domain"></application>
      <application app_name="answer" app_data=""></application>
      <application app_name="sleep" app_data="1000"></application>
      <application app_name="voicemail" app_data="default 
siplocal.my.domain ${dialed_ext}"></application>
    </extension>
    <caller_profile>
      <username>1002</username>
      <dialplan>XML</dialplan>
      <caller_id_name>Freeswitch1002</caller_id_name>
      <ani></ani>
      <aniii></aniii>
      <caller_id_number>1002</caller_id_number>
      <network_addr>192.168.178.1</network_addr>
      <rdnis></rdnis>
      <destination_number>1003</destination_number>
      <uuid>69a1337c-8a86-11dd-97c6-c70d0d682f22</uuid>
      <source>mod_sofia</source>
      <context>default</context>
      <chan_name>sofia/internal/1002@siplocal.my.domain</chan_name>
    </caller_profile>
    <times>
      <created_time>1222294579335872</created_time>
      <profile_created_time>1222294579335872</profile_created_time>
      <progress_time>0</progress_time>
      <progress_media_time>0</progress_media_time>
      <answered_time>1222294583270171</answered_time>
      <hangup_time>1222294586861282</hangup_time>
      <transfer_time>0</transfer_time>
    </times>
  </callflow>

Best regards
Peter

[Freeswitch-users] Cause: MANDATORY_IE_MISSING - No dial-string available

Peter P GMX