Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an [EMAIL PROTECTED] setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP software wouldn't work )
I've finally managed to get incoming calls to work properly by getting it to forward all calls to 4000 which is then passed on to the asterisk proxy and treated as an inbound route that gets answered correctly. The problem is then that when I place an outbound call through the gateway it also forwards that back as well. It then uses each channel in order until it fails as they're all busy. The xml configuration file is at http://www.ascensus.co.uk/config.xml The asterisk debug log is as below with my mobile replaced with mymobileno: I've also attached sip.conf below. If anyone has any idea how to get this thing to accept outgoing calls I would be very grateful of any input. All the docs and howto's I've found state that it should 'just work' once the inbound settings are working but I've not found that to be the case. The settings are all defaults except the following: Static IP address Proxy server address VAD on 711 disabled Comfort noise disabled AutomaticCallEnable yes AutomaticCallTargetAddress 4000 (which is obviously the problem...) Any help appreciated Thanks robbie Sip.conf <snip> [inbound] type=friend host=192.168.0.253 context=from-pstn canreinvite=no allow=ulaw allow=alaw <snip> asterisk1*CLI> -- Executing Macro("SIP/4005-9d61", "dialout-trunk|7|mymobilenumber|") in new stack -- Executing GotoIf("SIP/4005-9d61", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/4005-9d61", "record-enable|4005|OUT") in new stack -- Executing GotoIf("SIP/4005-9d61", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf("SIP/4005-9d61", "1?5:8") in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget("SIP/4005-9d61", "RecEnable=RECORD-OUT/4005") in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=4005 -- DBget: Value not found in database. -- Executing SetVar("SIP/4005-9d61", "CALLFILENAME=OUT4005-20070411-181258-1176311578.13302") in new stack -- Executing Goto("SIP/4005-9d61", "s|14") in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf("SIP/4005-9d61", "0?15:99") in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp("SIP/4005-9d61", "NO RECORDING NEEDED") in new stack -- Executing GotoIf("SIP/4005-9d61", "fooBgate:?7") in new stack -- Executing SetCallerID("SIP/4005-9d61", "Bgate: Treatment (Large) <4005>") in new stack -- Executing Goto("SIP/4005-9d61", "9") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing SetGroup("SIP/4005-9d61", "OUT_7") in new stack -- Executing CheckGroup("SIP/4005-9d61", "") in new stack -- Executing SetVar("SIP/4005-9d61", "DIAL_NUMBER=mymobilenumber") in new stack -- Executing SetVar("SIP/4005-9d61", "DIAL_TRUNK=7") in new stack -- Executing AGI("SIP/4005-9d61", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/4005-9d61", "OUTNUM=mymobilenumber") in new stack -- Executing Cut("SIP/4005-9d61", "custom=OUT_7|:|1") in new stack -- Executing GotoIf("SIP/4005-9d61", "0?19") in new stack -- Executing Dial("SIP/4005-9d61", "SIP/inbound/mymobilenumber") in new stack We're at 192.168.0.254 port 12542 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 From: "Bgate: Treatment (Large)" <sip:[EMAIL PROTECTED]>;tag=as5b17ec6a To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 11 Apr 2007 17:12:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 240 v=0 o=root 1321 1321 IN IP4 192.168.0.254 s=session c=IN IP4 192.168.0.254 t=0 0 m=audio 12542 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.0.253:5060 -- Called inbound/mymobilenumber asterisk1*CLI> Sip read: SIP/2.0 100 Trying Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: "Bgate: Treatment (Large)" <sip:[EMAIL PROTECTED]>;tag=as5b17ec6a To: <sip:[EMAIL PROTECTED]>;tag=2120bdca0a07567 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 Content-Length: 0 User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 8 headers, 0 lines asterisk1*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da Content-Length: 296 To: sip:[EMAIL PROTECTED] From: Incoming <sip:[EMAIL PROTECTED]>;tag=68b2ce27259fa46 Call-ID: [EMAIL PROTECTED] CSeq: 527193825 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: Incoming <sip:[EMAIL PROTECTED]> Supported: replaces User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 v=0 o=MxSIP 1616130847127821823 1523581411134402127 IN IP4 192.168.0.253 s=- c=IN IP4 192.168.0.253 t=0 0 a=sendrecv m=audio 5004 RTP/AVP 0 18 4 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 15 headers, 13 lines Using latest request as basis request Sending to 192.168.0.253 : 5060 (non-NAT) Found peer 'inbound' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 192.168.0.253:5004 Found description format PCMU Found description format G729 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 4000 in from-pstn list_route: hop: <sip:[EMAIL PROTECTED]> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da From: Incoming <sip:[EMAIL PROTECTED]>;tag=68b2ce27259fa46 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 527193825 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.253:5060 -- Executing SetVar("SIP/192.168.0.254-b2159358", "FROM_DID=4000") in new stack -- Executing Goto("SIP/192.168.0.254-b2159358", "aa_2|s|1") in new stack -- Goto (aa_2,s,1) -- Executing GotoIf("SIP/192.168.0.254-b2159358", "0?4") in new stack -- Executing Answer("SIP/192.168.0.254-b2159358", "") in new stack We're at 192.168.0.254 port 18268 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da From: Incoming <sip:[EMAIL PROTECTED]>;tag=68b2ce27259fa46 To: sip:[EMAIL PROTECTED];tag=as3f53dd1e Call-ID: [EMAIL PROTECTED] CSeq: 527193825 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 237 v=0 o=root 1321 1321 IN IP4 192.168.0.254 s=session c=IN IP4 192.168.0.254 t=0 0 m=audio 18268 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - to 192.168.0.253:5060 -- Executing Wait("SIP/192.168.0.254-b2159358", "1") in new stack asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK865a4fbe1 Content-Length: 0 To: sip:[EMAIL PROTECTED];tag=as3f53dd1e From: Incoming <sip:[EMAIL PROTECTED]>;tag=68b2ce27259fa46 Call-ID: [EMAIL PROTECTED] CSeq: 527193825 ACK Contact: Incoming <sip:[EMAIL PROTECTED]> User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 9 headers, 0 lines -- Executing SetVar("SIP/192.168.0.254-b2159358", "DIR-CONTEXT=general") in new stack -- Executing DigitTimeout("SIP/192.168.0.254-b2159358", "3") in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout("SIP/192.168.0.254-b2159358", "7") in new stack -- Set Response Timeout to 7 -- Executing BackGround("SIP/192.168.0.254-b2159358", "custom/aa_2") in new stack -- Playing 'custom/aa_2' (language 'en') asterisk1*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098 Content-Length: 294 To: sip:[EMAIL PROTECTED] From: Incoming <sip:[EMAIL PROTECTED]>;tag=c4f38b97a1b7d91 Call-ID: [EMAIL PROTECTED] CSeq: 1922136933 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: Incoming <sip:[EMAIL PROTECTED]> Supported: replaces User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 v=0 o=MxSIP 666448690115909283 852756300509302897 IN IP4 192.168.0.253 s=- c=IN IP4 192.168.0.253 t=0 0 a=sendrecv m=audio 5006 RTP/AVP 0 18 4 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 15 headers, 13 lines Using latest request as basis request Sending to 192.168.0.253 : 5060 (non-NAT) Found peer 'inbound' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 192.168.0.253:5006 Found description format PCMU Found description format G729 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 4000 in from-pstn list_route: hop: <sip:[EMAIL PROTECTED]> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098 From: Incoming <sip:[EMAIL PROTECTED]>;tag=c4f38b97a1b7d91 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1922136933 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.253:5060 -- Executing SetVar("SIP/192.168.0.254-b21601d0", "FROM_DID=4000") in new stack -- Executing Goto("SIP/192.168.0.254-b21601d0", "aa_2|s|1") in new stack -- Goto (aa_2,s,1) -- Executing GotoIf("SIP/192.168.0.254-b21601d0", "0?4") in new stack -- Executing Answer("SIP/192.168.0.254-b21601d0", "") in new stack We're at 192.168.0.254 port 13034 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098 From: Incoming <sip:[EMAIL PROTECTED]>;tag=c4f38b97a1b7d91 To: sip:[EMAIL PROTECTED];tag=as662b9545 Call-ID: [EMAIL PROTECTED] CSeq: 1922136933 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 237 v=0 o=root 1321 1321 IN IP4 192.168.0.254 s=session c=IN IP4 192.168.0.254 t=0 0 m=audio 13034 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - to 192.168.0.253:5060 -- Executing Wait("SIP/192.168.0.254-b21601d0", "1") in new stack asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bKcc0a3292d Content-Length: 0 To: sip:[EMAIL PROTECTED];tag=as662b9545 From: Incoming <sip:[EMAIL PROTECTED]>;tag=c4f38b97a1b7d91 Call-ID: [EMAIL PROTECTED] CSeq: 1922136933 ACK Contact: Incoming <sip:[EMAIL PROTECTED]> User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 9 headers, 0 lines -- Executing SetVar("SIP/192.168.0.254-b21601d0", "DIR-CONTEXT=general") in new stack -- Executing DigitTimeout("SIP/192.168.0.254-b21601d0", "3") in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout("SIP/192.168.0.254-b21601d0", "7") in new stack -- Set Response Timeout to 7 -- Executing BackGround("SIP/192.168.0.254-b21601d0", "custom/aa_2") in new stack -- Playing 'custom/aa_2' (language 'en') asterisk1*CLI> Sip read: SIP/2.0 480 Temporarily Unavailable Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: "Bgate: Treatment (Large)" <sip:[EMAIL PROTECTED]>;tag=as5b17ec6a To: <sip:[EMAIL PROTECTED]>;tag=2120bdca0a07567 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 Content-Length: 0 User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 8 headers, 0 lines -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.0.253 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 From: "Bgate: Treatment (Large)" <sip:[EMAIL PROTECTED]>;tag=as5b17ec6a To: <sip:[EMAIL PROTECTED]>;tag=2120bdca0a07567 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.253:5060 -- SIP/inbound-5c55 is circuit-busy == Everyone is busy/congested at this time -- Executing Goto("SIP/4005-9d61", "s-CONGESTION|1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp("SIP/4005-9d61", "Dial failed due to CONGESTION") in new stack -- Executing Macro("SIP/4005-9d61", "outisbusy") in new stack -- Executing Answer("SIP/4005-9d61", "") in new stack -- Executing Playtones("SIP/4005-9d61", "congestion") in new stack -- Executing Congestion("SIP/4005-9d61", "") in new stack Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa Content-Length: 295 To: sip:[EMAIL PROTECTED] From: Incoming <sip:[EMAIL PROTECTED]>;tag=13c1d178078adcb Call-ID: [EMAIL PROTECTED] CSeq: 375158022 INVITE Supported: timer Min-SE: 1800 Session-Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Contact: Incoming <sip:[EMAIL PROTECTED]> Supported: replaces User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 v=0 o=MxSIP 1766393978165944161 643256180583431736 IN IP4 192.168.0.253 s=- c=IN IP4 192.168.0.253 t=0 0 a=sendrecv m=audio 5008 RTP/AVP 0 18 4 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 15 headers, 13 lines Using latest request as basis request Sending to 192.168.0.253 : 5060 (non-NAT) Found peer 'inbound' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 192.168.0.253:5008 Found description format PCMU Found description format G729 Found description format G723 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 4000 in from-pstn list_route: hop: <sip:[EMAIL PROTECTED]> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa From: Incoming <sip:[EMAIL PROTECTED]>;tag=13c1d178078adcb To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 375158022 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.253:5060 -- Executing SetVar("SIP/192.168.0.254-b2167540", "FROM_DID=4000") in new stack -- Executing Goto("SIP/192.168.0.254-b2167540", "aa_2|s|1") in new stack -- Goto (aa_2,s,1) -- Executing GotoIf("SIP/192.168.0.254-b2167540", "0?4") in new stack -- Executing Answer("SIP/192.168.0.254-b2167540", "") in new stack We're at 192.168.0.254 port 11990 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa From: Incoming <sip:[EMAIL PROTECTED]>;tag=13c1d178078adcb To: sip:[EMAIL PROTECTED];tag=as4f5cc934 Call-ID: [EMAIL PROTECTED] CSeq: 375158022 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 237 v=0 o=root 1321 1321 IN IP4 192.168.0.254 s=session c=IN IP4 192.168.0.254 t=0 0 m=audio 11990 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - to 192.168.0.253:5060 -- Executing Wait("SIP/192.168.0.254-b2167540", "1") in new stack asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bKd6fd96da5 Content-Length: 0 To: sip:[EMAIL PROTECTED];tag=as4f5cc934 From: Incoming <sip:[EMAIL PROTECTED]>;tag=13c1d178078adcb Call-ID: [EMAIL PROTECTED] CSeq: 375158022 ACK Contact: Incoming <sip:[EMAIL PROTECTED]> User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 9 headers, 0 lines -- Executing SetVar("SIP/192.168.0.254-b2167540", "DIR-CONTEXT=general") in new stack -- Executing DigitTimeout("SIP/192.168.0.254-b2167540", "3") in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout("SIP/192.168.0.254-b2167540", "7") in new stack -- Set Response Timeout to 7 -- Executing BackGround("SIP/192.168.0.254-b2167540", "custom/aa_2") in new stack -- Playing 'custom/aa_2' (language 'en') == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/4005-9d61' in macro 'outisbusy' == Spawn extension (from-internal, 88mymobilenumber, 2) exited non-zero on 'SIP/4005-9d61' -- Executing Macro("SIP/4005-9d61", "hangupcall") in new stack -- Executing ResetCDR("SIP/4005-9d61", "w") in new stack -- Executing NoCDR("SIP/4005-9d61", "") in new stack -- Executing Wait("SIP/4005-9d61", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/4005-9d61' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4005-9d61' _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
