Mike, you answered: effective_caller_id_number is meant to be set on the a leg, not the b-leg. Well I need to control the "effective_caller_id_number" (or whatever it is called in the B-leg) individually for each location when I bridge to many destinations. In my case, I need to have different identities (A-numbers) for local calls and for external calls. I have followed the WIKI: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_ multiple_destinations, though. Anyhow how is this solved? Could it be possible to loopback to the dialplan as in Asterisk local channel or to go through some steps before dial attemps are made, but after the bridge application? Jon ------------------------------ Message: 4 Date: Fri, 19 Sep 2008 09:22:51 -0400 From: Michael Jerris <mike@jerris.com> Subject: Re: [Freeswitch-users] dialpaln To: freeswitch-users@lists.freeswitch.org Message-ID: <B017FCD2-1BF2-4EA3-83AA-786BC26BE141@jerris.com> Content-Type: text/plain; charset="us-ascii" On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote:

Hi,

Since I am not able to make the outbound call, when I use this command <oz dump 1 a> in the console, I used to get the all the 31 channels , for a refrence I am posing one channel block here,

span_id: 1 chan_id: 31 physical_span_id: 1 physical_chan_id: 31 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE

In this I am seeing that the state seems to be DOWN, but when I try wantouer status, it show as connected like ,

in openzap, state down is like on-hook... we should change that name maybe.

Wanrouter Status:

Device name | Protocol | Station | Status | wanpipe1 | AFT HDLC | N/A | Connected | wanpipe2 | AFT HDLC | N/A | Connecting |

where I am using is Sangoma A102D card with hardware echo cancellation. And when I try to dial from the softphone in the console I get this as,

freeswitch@dialer.optimus.co.in> 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1002@172.20.176.31 [a75028c4-554a-473e-a8cc-8c52d7f72df4] 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing FreeSwitch->99841799874 in context default 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel OpenZAP/1:1/9841799874 [bd977e0d-e078-4193-a005-d751bdb26db8] 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state() Hangup sofia/internal/1002@172.20.176.31 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321 switch_ivr_originate() Hangup OpenZAP/1:1/9841799874 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 1 (sofia/internal/1002@172.20.176.31 ) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1002@172.20.176.31 [CS_HANGUP] 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874 [CS_HANGUP]

So where i am wrong, can you please correct me.

It looks like the hang-up is coming from the sip side of things, can you take a look at the sip trace and confirm there is really a cancel on sip. Mike