[Asterisk-Users] asterisk and cisco call manager via h.323
Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party name: [500] -- Called party number: [500] -- Executing Playback(H323/ip$10.0.1.219:2303/8, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') =*= In CreateRealTimeLogicalChannel for call 8 -- externalIpAddress: 10.0.1.207 -- externalPort: 15210 -- SessionID: 1 -- Direction: IsTransmitter -- Started logical channel: sending G.711-uLaw-64k{sw} -- channelsOpen = 1 -- remoteIpAddress: 0.0.0.0 -- remotePort: 0 -- ExternalIpAddress: 10.0.1.207 -- ExternalPort: 15210 -- Connection Established with 5001, 5001 [10.0.1.219] =*= In CreateRealTimeLogicalChannel for call 8 -- externalIpAddress: 10.0.1.207 -- externalPort: 15210 -- SessionID: 1 -- Direction: IsReceiver -- Started logical channel: receiving G.711-uLaw-64k{sw} -- channelsOpen = 2 -- remoteIpAddress: 10.0.1.219 -- remotePort: 4000 -- ExternalIpAddress: 10.0.1.207 -- ExternalPort: 15210 -- remoteIpAddress: 0.0.0.0 remotePort: 0 Looks incorrectly ! Tested with latest cvs asterisk. Maybe asterisk h.323 channel driver not correctly parse h.323 messages. -- Pavel Zheltouhov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
Thomas Dingermann wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? Is it possible to do this hack in chan_sip? I think it's too dificult for me ) Threewaycalling is the only thing i am missing with ATA/SIP/Asterisk! -or- does ATA/MGCP/Asterisk complete working (CallerID-transfer, No, as i know. MSG-Waiting-Indicator...)? Maybe. Can you post a chan_mgcp.diff or your modified chan_mgcp? I really like to use threewaycalling with my ATAs. it's simple : 1539a1540,1545 if (strpbrk(tone,0123456789*#)) { if (mgcpdebug) { ast_verbose(VERBOSE_PREFIX_3 MGCP Asked to indicate filtered tone,cisco workaround enabled \n); } return 0; } works for me (tm) You need cvs version, 0.4 does not work with flashhook messages at all. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mgcp problems
I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 306 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 307 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 308 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- Executing Playback(MGCP/aaln/[EMAIL PROTECTED], demo-echotest) in new stack -- Modified aaln/[EMAIL PROTECTED] with new mode: sendrecv on callid: 7d4b8e932149c6df Posting Request: MDCX 309 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 7d4b8e932149c6df What is the -1 'UNKNOWN' condition on channel ? Is it correct mgcp packet ? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] three way calling and cisco ata 186
Thomas Dingermann wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? I just try two ATA with asterisk with that configuration files : ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 0.0.0.0 allow=ulaw inbanddtmf=yes transfer = yes threewaycalling=yes [10.0.1.19] transfer = yes threewaycalling=yes host = 10.0.1.19 context = default line = aaln/1 transfer = 1 line = aaln/2 transfer = 1 line = * [10.0.1.20] transfer = yes threewaycalling=yes host = 10.0.1.20 context = default line = aaln/1 transfer = 1 line = aaln/2 transfer = 1 line = * and extensions.conf --- exten = 31,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 32,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 33,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 34,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) - Ordinary tasks works good. Call transfer with '#' key work too. But three way calling not work with stange error : -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down dial to 33 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3' -- Executing Dial(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED]|20|tr) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing answer -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered MGCP/aaln/[EMAIL PROTECTED] -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] Talking now Attempt call person 3 : hookflash : -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down now trying dial to other phone ( 600 - echo test ) -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '6' atfter this, person1 hear 'fastbusy', short beeps ! And other output of asterisk: -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- Executing Playback(MGCP/aaln/[EMAIL PROTECTED], demo-echotest) in new stack -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Playing 'demo-echotest' -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] -- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED] -- MGCP Muting 0 on aaln/[EMAIL PROTECTED] -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED]MASQ) WARNING[20501]: File chan_mgcp.c, Line 764 (mgcp_fixup): old channel wasn't 0x81065a8 but was (nil) WARNING[20501]: File channel.c, Line 1847 (ast_do_masquerade): Fixup failed on channel MGCP/aaln/[EMAIL PROTECTED]MASQ, strange things may happen. NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED]ZOMBIE, MGCP/aaln/[EMAIL PROTECTED]) -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' == Spawn extension (default, 600, 1) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]' -- Any ideas ? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] three way calling and cisco ata 186
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is enabled by setting bits 28-29 to 10 ( Select the Cisco VG248 Style for mid-call services. as described at http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4d1f.html#42433 But it seems not works,I always get conference call with 3 persons. -- Pavel Zheltouhov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users