From owner-freebsd-isdn@FreeBSD.ORG Mon Sep 3 21:18:54 2007 Return-Path: Delivered-To: freebsd-isdn@FreeBSD.org Received: from mx1.freebsd.org (mx1.freebsd.org [IPv6:2001:4f8:fff6::34]) by hub.freebsd.org (Postfix) with ESMTP id CA00D16A41A for ; Mon, 3 Sep 2007 21:18:54 +0000 (UTC) (envelope-from ovb@ovb.ch) Received: from ovbis01.ovb.ch (ovbis01.ovb.ch [213.188.32.144]) by mx1.freebsd.org (Postfix) with ESMTP id 7708013C480 for ; Mon, 3 Sep 2007 21:18:54 +0000 (UTC) (envelope-from ovb@ovb.ch) Received: from ovbas00.ovb.ch ([213.180.173.192] helo=[192.168.30.103]) by ovbis01.ovb.ch with esmtp (Exim 4.51) id 1ISJJt-0000zv-8Y; Mon, 03 Sep 2007 23:18:53 +0200 Message-ID: <46DC7A3D.8000109@ovb.ch> Date: Mon, 03 Sep 2007 23:18:53 +0200 From: Oliver von Bueren User-Agent: Thunderbird 2.0.0.6 (Windows/20070728) MIME-Version: 1.0 To: freebsd-isdn@FreeBSD.org Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Cc: Subject: Asterisk with ISDN4BSD X-BeenThere: freebsd-isdn@freebsd.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: Using ISDN with FreeBSD List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , X-List-Received-Date: Mon, 03 Sep 2007 21:18:55 -0000 Hi M8s I've a strange problem which I just started to notice. I've connected my asterisk box to a P2P ISDN line (telco provider) with a FritzCard. Running the i4b by hps on a FreeBSD 6.2-p7 It seems that on every second INCOMING call, the audio signal is not routed from the ISDN to the SIP phone. Outgoing is ok. In this example, it worked, calling from an external number to DDI 41 -- capi_handle_connect_indication:6049:ENTRY=:PLCI=0x0300:PBX_CHAN=**Unknown**: -- Incoming call from '079xxxxxxxx' to '41', CIP=0x0010, sending_complete=yes == search_cep:5818:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=**Unknown**: == Incoming call '079xxxxxxxx' -> '41' == cd_start_pbx:5939:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: == Started PBX -- Executing [41@from_isdn:1] Set("CAPI/ISDN1/41-2", "CALLERID(num)=0079xxxxxxx") in new stack -- Executing [41@from_isdn:2] Goto("CAPI/ISDN1/41-2", "to_sip|705|1") in new stack -- Goto (to_sip,705,1) -- Executing [705@to_sip:1] Ringing("CAPI/ISDN1/41-2", "") in new stack == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: -- Executing [705@to_sip:2] Dial("CAPI/ISDN1/41-2", "SIP/705|30") in new stack -- Called 705 > Out of order update usecount! -- SIP/705-087a8000 is ringing == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: -- SIP/705-087a8000 is ringing -- SIP/705-087a8000 answered CAPI/ISDN1/41-2 == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: == chan_capi_answer:4702:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: == capi_send_connect_resp:3128:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: == Connected to 41 == capi_send_detect_dtmf_req:3449:ENTRY=ISDN1:PLCI=0x0300:PBX_CHAN=CAPI/ISDN1/41-2: == Setting up DTMF detector, flag=1 [...] => call takes place [...] == Spawn extension (to_sip, 705, 2) exited non-zero on 'CAPI/ISDN1/41-2' > Out of order update usecount! ------------------------------------------------------------------------------------------ Right after the call above, I did the next one, which does not work: ------------------------------------------------------------------------------------------ -- capi_handle_connect_indication:6049:ENTRY=:PLCI=0x0400:PBX_CHAN=**Unknown**: -- Incoming call from '079xxxxxxx' to '41', CIP=0x0010, sending_complete=yes == search_cep:5818:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=**Unknown**: == Incoming call '079xxxxxxx' -> '41' == cd_start_pbx:5939:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: == Started PBX -- Executing [41@from_isdn:1] Set("CAPI/ISDN1/41-3", "CALLERID(num)=0079xxxxxxx") in new stack -- Executing [41@from_isdn:2] Goto("CAPI/ISDN1/41-3", "to_sip|705|1") in new stack -- Goto (to_sip,705,1) -- Executing [705@to_sip:1] Ringing("CAPI/ISDN1/41-3", "") in new stack == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: -- Executing [705@to_sip:2] Dial("CAPI/ISDN1/41-3", "SIP/705|30") in new stack -- Called 705 > Out of order update usecount! -- SIP/705-087af000 is ringing == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: -- SIP/705-087af000 is ringing -- SIP/705-087af000 answered CAPI/ISDN1/41-3 == chan_capi_indicate:4643:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: == chan_capi_answer:4702:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: == capi_send_connect_resp:3128:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: == Connected to 41 == capi_send_detect_dtmf_req:3449:ENTRY=ISDN1:PLCI=0x0400:PBX_CHAN=CAPI/ISDN1/41-3: == Setting up DTMF detector, flag=1 [... call - hangup ...] == Spawn extension (to_sip, 705, 2) exited non-zero on 'CAPI/ISDN1/41-3' > Out of order update usecount! ------------------------------------------------------------------------------------------ I can repeat that multiple times, every second inbound call has no audio. What is interesting is, that the calls with an odd PLCI number (PLCI=0x0300 in the first log part) do work, the ones with an even number like PLCI=0x0400 do not. The Dialplan part for this call: [from_isdn] exten => 41,1,Set(CALLERID(num)=0${CALLERID(num)}) exten => 41,n,Goto(to_sip,705,1) [to_sip] exten => _70[1-3],1,Ringing exten => _70[1-3],n,Dial(SIP/${EXTEN},120) exten => _70[1-3],n,Hangup -------- capi.conf [general] nationalprefix=0 internationalprefix=00 rxgain=1.0 txgain=1.0 [ISDN1] isdnmode=msn incomingmsn=* controller=0 group=1 softdtmf=off relaxdtmf=off accountcode= context=from_isdn holdtype=local echocancel=no echosquelch=yes devices=2 wait_silence_samples=1000 --------------- # isdnconfig controller 0 = { Layer 1: description : AVM Fritz!Card PCI type : passive ISDN (Basic Rate, 2xB) channels : 0x3 serial : 0xabcd power_save : on dialtone : enabled attached : yes PH-state : Activate indication (priority=8/9) Layer 2: driver_type : DRVR_DSS1_P2P_TE } ------ Any ideas what that could be? -- Oliver