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[Prometheus001 at gmx.net]

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 ([email]sofia/internal/1002@siplocal.my.domain[/email]) 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 ([email]sofia/internal/1002@siplocal.my.domain[/email]) 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 ([email]sofia/internal/1002@siplocal.my.domain[/email]) Execute bridge(user/${dialed_ext}@siplocal.my.domain ([email]dialed_ext}@siplocal.my.domain[/email]))
2008-09-24 23:30:55 [DEBUG] switch_core_session.c:1053 switch_core_session_execute_application() sofia/internal/1002@siplocal.my.domain ([email]sofia/internal/1002@siplocal.my.domain[/email]) Expanded String bridge(user/1003@siplocal.my.domain ([email]user/1003@siplocal.my.domain[/email]))
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" (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" ([email]user/1003@siplocal.my.domain[/email])></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" ([email]user/@siplocal.my.domain[/email])></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 ([email]sofia/internal/1002@siplocal.my.domain[/email])</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 ([email]1003@192.168.178.1:2127;transport=tls;fs_nat=yes[/email])</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 ([email]sofia/internal/1003@192.168.178.1:2127;transport=tls;fs_nat=yes[/email])</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 ([email]1003@192.168.178.1:2127;transport=tls;fs_nat=yes[/email])</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 ([email]sofia/internal/1003@192.168.178.1:2127;transport=tls;fs_nat=yes[/email])</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" ([email]user/1003@siplocal.my.domain[/email])></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" ([email]user/@siplocal.my.domain[/email])></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 ([email]sofia/internal/1002@siplocal.my.domain[/email])</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

[Prometheus001 at gmx.net]

I figured out (via ngrep) that freeswitch didn't even try to contact the
registered phone. However it does a lookup to the Directory of the
target phone. So there may be something wrong with the bridge command?

BTW: To be honest: I haven't really understood the "user" part in the
bridge parameter in this case:
user/${dialed_ext}@siplocal.my.domain)
Where does "user" come from? Is it an implicit internal context?
May I use another context in that case when the phone is registered in
the internal context?

Best regards
Peter

Peter P GMX schrieb:

[mike at jerris.com]

On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:

[Prometheus001 at gmx.net]

Hello Michael,

thanks for the hint, but how shall a dial-string param look like? I
looked up the internet but could not find an example.
Can you provide an example?

Best regards
Peter

Michael Jerris schrieb:

[mike at jerris.com]

On Sep 25, 2008, at 4:04 PM, Peter P GMX wrote:

[brian at freeswitch.org]

I also must point out the user endpoint is there to normalize your
dial plan. because user/1000@domain.com could translate into an
openzap channel. while user/1001@domain.com could translate into a
sofia endpoint. The user param dial-string is used over the domain
level dial-string.

And with the addition of mod_loopback you could have user/1003@domain.com
translate into Loopback/1003@context ... and you have no idea its
happening on the under side.

/b

On Sep 25, 2008, at 1:04 PM, Peter P GMX wrote:

[Prometheus001 at gmx.net]

Thanks for the hint. With the following dial-string in the directory it
works:

<param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>

Best regards
Peter





Michael Jerris schrieb: