[Freeswitch-users] Unable to set internal call to registered sip user
Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : include domain name=$${domain} user id=22 mailbox=22 params param name=password value=Xk21%/param param name=vm-password value=22/param param name=sip-port value=5060/param /params variables variable name=accountcode value=22/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 22/variable variable name=effective_caller_id_number value=22/variable /variables /user user id=24 mailbox=24 params param name=password value=dudeldum/param param name=vm-password value=24/param param name=sip-port value=5060/param /params variables variable name=accountcode value=24/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 24/variable variable name=effective_caller_id_number value=24/variable /variables /user /domain /include It seems, that they can connect to the freeswitch. I configured the dialplan like following : include context name=default extension name=diallocal condition field=destination_number expression=^(2[0-9])$ !--- The % behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extensi...@${domain_name}/action !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- /condition /extension ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24-22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [...@192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/2...@192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _ Filip Lyncker, Dipl.-Inform. (FH) Lyncker Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Unable to set internal call to registered sip user
and this is not enough for you? !--- The *%* behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extension}@ ${domain_name}/ !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- T. On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de wrote: Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : include domain name=$${domain} user id=22 mailbox=22 params param name=password value=Xk21%/param param name=vm-password value=22/param param name=sip-port value=5060/param /params variables variable name=accountcode value=22/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 22/variable variable name=effective_caller_id_number value=22/variable /variables /user user id=24 mailbox=24 params param name=password value=dudeldum/param param name=vm-password value=24/param param name=sip-port value=5060/param /params variables variable name=accountcode value=24/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 24/variable variable name=effective_caller_id_number value=24/variable /variables /user /domain /include It seems, that they can connect to the freeswitch. I configured the dialplan like following : include context name=default extension name=diallocal condition field=destination_number expression=^(2[0-9])$ !--- The % behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extensi...@${domain_name}/action !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- /condition /extension ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24-22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [...@192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/2...@192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _ Filip Lyncker, Dipl.-Inform. (FH) Lyncker Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Unable to set internal call to registered sip user
ok , i tried several things : action application=bridge data=user/${dialed_extension}%${domain_name}/action action application=bridge data=user/${dialed_extension}%192.168.1.34/action action application=bridge data=user/${dialed_extension}/action but all this doesnt work sorry mybe I dont see something apparent , but I dont have a clue... Tihomir Culjaga schrieb: and this is not enough for you? !--- The *%* behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extensi...@${domain_name}/ !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- T. On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de mailto:lync...@lyth.de wrote: Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : include domain name=$${domain} user id=22 mailbox=22 params param name=password value=Xk21%/param param name=vm-password value=22/param param name=sip-port value=5060/param /params variables variable name=accountcode value=22/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 22/variable variable name=effective_caller_id_number value=22/variable /variables /user user id=24 mailbox=24 params param name=password value=dudeldum/param param name=vm-password value=24/param param name=sip-port value=5060/param /params variables variable name=accountcode value=24/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 24/variable variable name=effective_caller_id_number value=24/variable /variables /user /domain /include It seems, that they can connect to the freeswitch. I configured the dialplan like following : include context name=default extension name=diallocal condition field=destination_number expression=^(2[0-9])$ !--- The % behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extensi...@${domain_name}/action !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- /condition /extension ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24-22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [...@192.168.1.34 http://192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _ Filip Lyncker, Dipl.-Inform. (FH) Lyncker Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis ___ FreeSWITCH-users mailing list
Re: [Freeswitch-users] Unable to set internal call to registered sip user
Hi in dialplan i see: condition field=destination_number expression=^(2[0-9])$ - check on variable destination_number and later action application=bridge data=user/${dialed_extensi...@${domain_name}/action - bridge to variable dialed_extension , other then checked destination_number or $1 from regexp try: action application=bridge data=user/${destination_number}%${sip_profile}/action or action application=bridge data=user/$1%${sip_profile}/action By Kleo On Tue, 22 Sep 2009, Filip Lyncker wrote: Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : include domain name=$${domain} user id=22 mailbox=22 params param name=password value=Xk21%/param param name=vm-password value=22/param param name=sip-port value=5060/param /params variables variable name=accountcode value=22/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 22/variable variable name=effective_caller_id_number value=22/variable /variables /user user id=24 mailbox=24 params param name=password value=dudeldum/param param name=vm-password value=24/param param name=sip-port value=5060/param /params variables variable name=accountcode value=24/variable variable name=user_context value=default/variable variable name=effective_caller_id_name value=Extension 24/variable variable name=effective_caller_id_number value=24/variable /variables /user /domain /include It seems, that they can connect to the freeswitch. I configured the dialplan like following : include context name=default extension name=diallocal condition field=destination_number expression=^(2[0-9])$ !--- The % behind the username tells FS to lookup the user in it's local sip_registration database -- action application=bridge data=user/${dialed_extensi...@${domain_name}/action !--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device -- !--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the @ instead of %: action application=bridge data=sofia/profilename/5...@x.x.x.x/ -- /condition /extension ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24-22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [...@192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/2...@192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _ | You have moved the mouse. # | Windows must be restarted for the changes to take effect. # | OK # ##/ ~~ ~~ ~~ ~~ ~~ ~~ ~~ Vladimir `KLEO' Klejch Kleo'at'netbox.cz ... ... ... ... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org