Cause: MANDATORY_IE_MISSING - No dial-string available
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 ${dialed_ext}"></application> </extension> <caller_profile> <username>1002</username> <dialplan>XML</dialplan> <caller_id_name>Freeswitch1002</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>1002</caller_id_number> <network_addr>192.168.178.1</network_addr> <rdnis></rdnis> <destination_number>1003</destination_number> <uuid>69a1337c-8a86-11dd-97c6-c70d0d682f22</uuid> <source>mod_sofia</source> <context>default</context> <chan_name>sofia/internal/1002@siplocal.my.domain</chan_name> </caller_profile> <times> <created_time>1222294579335872</created_time> <profile_created_time>1222294579335872</profile_created_time> <progress_time>0</progress_time> <progress_media_time>0</progress_media_time> <answered_time>1222294583270171</answered_time> <hangup_time>1222294586861282</hangup_time> <transfer_time>0</transfer_time> </times> </callflow> Best regards Peter
I figured out (via ngrep) that freeswitch didn't even try to contact the registered phone. However it does a lookup to the Directory of the target phone. So there may be something wrong with the bridge command? BTW: To be honest: I haven't really understood the "user" part in the bridge parameter in this case: user/${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:
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 ${dialed_ext}"></application> </extension> <caller_profile> <username>1002</username> <dialplan>XML</dialplan> <caller_id_name>Freeswitch1002</caller_id_name> <ani></ani> <aniii></aniii> <caller_id_number>1002</caller_id_number> <network_addr>192.168.178.1</network_addr> <rdnis></rdnis> <destination_number>1003</destination_number> <uuid>69a1337c-8a86-11dd-97c6-c70d0d682f22</uuid> <source>mod_sofia</source> <context>default</context> <chan_name>sofia/internal/1002@siplocal.my.domain</chan_name> </caller_profile> <times> <created_time>1222294579335872</created_time> <profile_created_time>1222294579335872</profile_created_time> <progress_time>0</progress_time> <progress_media_time>0</progress_media_time> <answered_time>1222294583270171</answered_time> <hangup_time>1222294586861282</hangup_time> <transfer_time>0</transfer_time> </times> </callflow>
Best regards Peter
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On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:
I figured out (via ngrep) that freeswitch didn't even try to contact the registered phone. However it does a lookup to the Directory of the target phone. So there may be something wrong with the bridge command?
BTW: To be honest: I haven't really understood the "user" part in the bridge parameter in this case: user/${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
user is a proxy endpoint (not a context) that looks up the user and uses the dial-string param from the user to make the call. You are missing that setting on the user, so we have no idea what to call. Mike
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]*
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:
On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:
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/${dialed_ext}@siplocal.my.domain) Where does "user" come from? Is it an implicit internal context? May I use another context in that case when the phone is registered in the internal context?
Best regards Peter
user is a proxy endpoint (not a context) that looks up the user and uses the dial-string param from the user to make the call. You are missing that setting on the user, so we have no idea what to call.
Mike
2008-09-24 23:30:56 [ERR] mod_dptools.c:1957 user_outgoing_channel() *No dial-string available, please check your user directory.* 2008-09-24 23:30:56 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: *[MANDATORY_IE_MISSING]* 2008-09-24 23:30:56 [DEBUG] switch_ivr_originate.c:1404 switch_ivr_originate() Originate Resulted in *Error Cause: 96 [MANDATORY_IE_MISSING]*
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On Sep 25, 2008, at 4:04 PM, Peter P GMX wrote:
Hello Michael,
thanks for the hint, but how shall a dial-string param look like? I looked up the internet but could not find an example. Can you provide an example?
Best regards Peter
its just an originate string like you use with bridge app or the originate FSAPI command. In the default configs we have it just as the default in the domain, but the same setting is good in the user: http://svn.freeswitch.org/svn/freeswitch/trunk/conf/directory/default.xml MIke
Thanks for the hint. With the following dial-string in the directory it works: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> Best regards Peter Michael Jerris schrieb:
On Sep 25, 2008, at 4:04 PM, Peter P GMX wrote:
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
its just an originate string like you use with bridge app or the originate FSAPI command. In the default configs we have it just as the default in the domain, but the same setting is good in the user:
http://svn.freeswitch.org/svn/freeswitch/trunk/conf/directory/default.xml
MIke
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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:
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
participants (3)
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Brian West
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Michael Jerris
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Peter P GMX