Re: [Asterisk-Users] optimizing for via C3
$(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY +CFLAGS:=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY CFLAGS+=$(OPTIMIZE) ifneq ($(PROC),ultrasparc) and set 'PROC=c3' instead of 'PROC=i386' as originally set by Debian in the master makefile (debian/rules) . Random inspection at build time gave me the impression that all binaries were built with -march=c3 . Yet, no improvement at all. As before, the generated debs and/or source debs/patch are availble upon request for you to re-run this. The system is Debian Sarge: ii libc6 2.3.2.ds1-22 GNU C Library: Shared libraries and Timezone ii gcc3.3.5-3The GNU C compiler ii libgsm11.0.10-13 Shared libraries for GSM speech compressor ii libspeex1 1.1.6-2The Speex Speech Codec Am I looking at the wrong numbers? If so: what do those numbers mean? Can anybody suggest useful and simple benchmarks? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
On linux raid: Linux raid supports hot swapping well. It doesn't care about the hardware, which is being swapped, much. Obviously, in simple disk scenario, which is used fot sw raid, only scsi and SATA can be hot-swapped. Also, make sure that the motherboard supports hot-swap SATA, i've seen some that have stickers that they don't, i can only guess how many don't put the stickers when they should. Also, linux raid performance is very good. HW raid gains perfromance boost because of extra cache they have onboard, thus peak writes are easily swallowed by cache and written when possible. As an end note, don't try to boot your linux raid with one or more hard drives missing, it will fail. If you remove the disk, make sure you put something back AND make sure you have the same partitions there. SATA is fast enough. In fact, ATAPI is also fast enough in most scenarios. It is just that SCSI disks/arrays tend to be of better quality (but usually much more expensive). IIRC Linux's raid support will support hot-swapping disks, but I'm not sure which disks are are supported. An external array with its own CPU doesn't necessarily mean better performance than one using the host CPU, BTW. Though it will take some load off of Asterisk. And if this is just about redundnacy and not about performance, consider not buying an expensive array at all, and using two cheap systems. The cost will be roughly the same, I believe. (RAID= Redundant Array of Inexpensive Disks). Any simple way to achive redundancy here? -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz, mISDN, Help
Haven't tried. The install scripts gets May's release and compiles with that. I think some serious porting will be nescessary. Anyone? On 10/09/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Sep 10, 2005 at 01:25:35PM +0300, Konrads Smelkovs wrote: Isn't billion a HFC PCI card? see lspci output, if so, use bristuff from junghanns.net http://www.junghanns.net/en/download.html , i suggest CVS version Does the CVS version (made sometime on May) still build with current HEAD? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz, mISDN, Help
Unfourtunatley, mISDN is far from production quality. So going miSDN-CAPI-chan_capi might not work. chan_misdn is even more flakey at the moment. Your best option is to use just CAPI and chan_capi,it had support for fritz On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote: A plea to all! Has anyone had any success with two or more avm fritz pci cards with either misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x? I have managed to get misdn to load under 2.6.13 and detect two cards using misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but the second card/controller doesn't answer or dial calls. But if I try misdn without capi I get the following error mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596 Any help would be greatly appreciated. Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz, mISDN, Help
Isn't billion a HFC PCI card? see lspci output, if so, use bristuff from junghanns.net http://www.junghanns.net/en/download.html , i suggest CVS version On 10/09/05, John Fawcett [EMAIL PROTECTED] wrote: Jon Dean wrote: A plea to all! Has anyone had any success with two or more avm fritz pci cards with either misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x? I have managed to get misdn to load under 2.6.13 and detect two cards using misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but the second card/controller doesn't answer or dial calls. But if I try misdn without capi I get the following error mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596 I have two Billions PCI cards and am using chan_misdn-0.1.1 and kernel 2.6.11. I haven't yet been successful at using the second card, but it's still work in progress. However, I got the above error and resolved it by specifying the parameters for both cards when I loaded the relevant driver. In my case: modprobe hfcpci protocol=0x2,0x12 layermaster=0xf,0x3 (the first card is in TE mode the second in NT mode) Hope this helps. John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can get IAX connection but no SIP connection?
did you enable sip debug? do you run asterisk with -vgc ? do you see packets arriving with tcpdump? On 06/09/05, Tim P [EMAIL PROTECTED] wrote: Ok I have a box with 2 nics, one outside and one inside (192.168.8.x) I want to get connections for sip and iax on the outside connection I have a firewall on the box but have opened the followng ports pbx 5060-5070 udp + tcp pbx 4569 udp pbx 5036 tcp pbx 2727 udp pbx 1-2 udp pbx 69 udp This allows me to connect with iax but sip fails registration with a 408. I watch the console and don't see a registration reuqest come in. I also made an entry in sip.conf to for bindaddr = myoutsideip Any idea of what I am missing? I see nothing in the logs pretaining to my sip connection. I am running asterisk 1.0.9 with verbosity 5 in my logs. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working
Here you go, eagerly awaiting comments: -- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack -- Executing Dial(SIP/xlite1-e0a7, CAPI/hfcpci/17/b) in new stack data = hfcpci/17/b parsed dialstring: 'hfcpci' '17' 'b' capi request for interface 'hfcpci' parsed dialstring: 'hfcpci' '17' 'b' == hfcpci: Call CAPI/hfcpci/17-0 with B3 (pres=0x00, ton=0x00) CONNECT_CONF ID=001 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- hfcpci: received CONNECT_CONF PLCI = 0x101 CONNECT_REQ ID=001 #0x0003 LEN=0044 Controller/PLCI/NCCI= 0x1 CIPValue= 0x1 CalledPartyNumber = 8017 CallingPartyNumber = 00 800 CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default -- Called hfcpci/17/b INFO_IND ID=001 #0x0001 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8 InfoElement = 81 81 INFO_RESP ID=001 #0x0001 LEN=0012 Controller/PLCI/NCCI= 0x101 -- hfcpci: info element CAUSE 81 81 DISCONNECT_IND ID=001 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 DISCONNECT_IND ID=001 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3481 DISCONNECT_RESP ID=001 #0x0002 LEN=0012 Controller/PLCI/NCCI= 0x101 CAPI INFO 0x3481: Unallocated (unassigned) number == hfcpci: CAPI Hangingup == hfcpci: Interface cleanup PLCI=0x101 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-e0a7' status is 'NOANSWER' On 04/09/05, Armin Schindler [EMAIL PROTECTED] wrote: This is not enough to see the problem. Use verbose level 5 (-v) and use 'capi debug' Armin On Sun, 4 Sep 2005, Konrads Smelkovs wrote: See if this helps... , i ran asterisk with -vvvgc CAPI Debugging Enabled -- Executing SetCallerID(SIP/xlite1-1be2, 0) in new stack -- Executing Dial(SIP/xlite1-1be2, CAPI/hfcpci/17/b) in new stack data = hfcpci/17/b capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/17-1 with B3 (pres=0x00, ton=0x00) -- hfcpci: received CONNECT_CONF PLCI = 0x101 -- Called hfcpci/17/b == hfcpci: Interface cleanup PLCI=0x101 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-1be2' status is 'NOANSWER' Maybe there is something more I could look for? On 02/09/05, Armin Schindler [EMAIL PROTECTED] wrote: On Fri, 2 Sep 2005, Konrads Smelkovs wrote: Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack -- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack data = hfcpci/b17 capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00, ton=0x00) -- hfcpci: received CONNECT_CONF PLCI = 0x201 -- Called hfcpci/b17 == hfcpci: Interface cleanup PLCI=0x201 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-3b80' status is 'NOANSWER' mISDNUser test tools show ISDN line working (testcon). capi info shows that 2 B channels are available capiinfo utility also dumps meaningful information - indicating that it indeed recognises the card. To see more, you may want to increase verbosity level and enable 'capi debug'. Anyway, if you are using CVS version of chan_capi, your dialstring is not correct. The option for earlyb3 'b' may not be part of the called number any more. Option are added after the called id and an additional '/'. Your dial command should look like this: Dial(CAPI/hfcpci/17/b) chan_capi version - CVS as of Sep 2, 2005 from chan-capi.sf.net I read comments on voip-info about 2.6.12 kernel breaking something, but the patch was for capi 0.3.5, not sure it applies... No, this is obsolete for chan_capi on sourceforge. No patches are needed. Armin -- Konrads Smelkovs Applied IT sorcery. -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth
Re: [Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working
Hello, I solved the problem - i was setting wrong caller-ID and thus got rejected. Thanks for help. On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote: Here you go, eagerly awaiting comments: -- Executing SetCallerID(SIP/xlite1-e0a7, 0) in new stack -- Executing Dial(SIP/xlite1-e0a7, CAPI/hfcpci/17/b) in new stack data = hfcpci/17/b parsed dialstring: 'hfcpci' '17' 'b' capi request for interface 'hfcpci' parsed dialstring: 'hfcpci' '17' 'b' == hfcpci: Call CAPI/hfcpci/17-0 with B3 (pres=0x00, ton=0x00) CONNECT_CONF ID=001 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- hfcpci: received CONNECT_CONF PLCI = 0x101 CONNECT_REQ ID=001 #0x0003 LEN=0044 Controller/PLCI/NCCI= 0x1 CIPValue= 0x1 CalledPartyNumber = 8017 CallingPartyNumber = 00 800 CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default -- Called hfcpci/17/b INFO_IND ID=001 #0x0001 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8 InfoElement = 81 81 INFO_RESP ID=001 #0x0001 LEN=0012 Controller/PLCI/NCCI= 0x101 -- hfcpci: info element CAUSE 81 81 DISCONNECT_IND ID=001 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 DISCONNECT_IND ID=001 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3481 DISCONNECT_RESP ID=001 #0x0002 LEN=0012 Controller/PLCI/NCCI= 0x101 CAPI INFO 0x3481: Unallocated (unassigned) number == hfcpci: CAPI Hangingup == hfcpci: Interface cleanup PLCI=0x101 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-e0a7' status is 'NOANSWER' On 04/09/05, Armin Schindler [EMAIL PROTECTED] wrote: This is not enough to see the problem. Use verbose level 5 (-v) and use 'capi debug' Armin On Sun, 4 Sep 2005, Konrads Smelkovs wrote: See if this helps... , i ran asterisk with -vvvgc CAPI Debugging Enabled -- Executing SetCallerID(SIP/xlite1-1be2, 0) in new stack -- Executing Dial(SIP/xlite1-1be2, CAPI/hfcpci/17/b) in new stack data = hfcpci/17/b capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/17-1 with B3 (pres=0x00, ton=0x00) -- hfcpci: received CONNECT_CONF PLCI = 0x101 -- Called hfcpci/17/b == hfcpci: Interface cleanup PLCI=0x101 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-1be2' status is 'NOANSWER' Maybe there is something more I could look for? On 02/09/05, Armin Schindler [EMAIL PROTECTED] wrote: On Fri, 2 Sep 2005, Konrads Smelkovs wrote: Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack -- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack data = hfcpci/b17 capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00, ton=0x00) -- hfcpci: received CONNECT_CONF PLCI = 0x201 -- Called hfcpci/b17 == hfcpci: Interface cleanup PLCI=0x201 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'SIP/xlite1-3b80' status is 'NOANSWER' mISDNUser test tools show ISDN line working (testcon). capi info shows that 2 B channels are available capiinfo utility also dumps meaningful information - indicating that it indeed recognises the card. To see more, you may want to increase verbosity level and enable 'capi debug'. Anyway, if you are using CVS version of chan_capi, your dialstring is not correct. The option for earlyb3 'b' may not be part of the called number any more. Option are added after the called id and an additional '/'. Your dial command should look like this: Dial(CAPI/hfcpci/17/b) chan_capi version - CVS as of Sep 2, 2005 from chan-capi.sf.net I read comments on voip-info about 2.6.12 kernel breaking something, but the patch
[Asterisk-Users] asterisk CAPI dial-in issues
Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten = _XX,1,Playback(demo-abouttotry) exten = _XX,n,Dial,SIP/xlite1 exten = _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until caller gives up: INFO_IND ID=001 #0x040a LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x7e InfoElement = 04 CAPI: no interface for PLCI = 0x101 MN = 0x40a INFO_RESP ID=001 #0x040a LEN=0012 Controller/PLCI/NCCI= 0x101 CAPI: INFO_IND no interface for PLCI=0x101 INFO_IND ID=001 #0x040b LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 CAPI: no interface for PLCI = 0x101 MN = 0x40b INFO_RESP ID=001 #0x040b LEN=0012 Controller/PLCI/NCCI= 0x101 CAPI: INFO_IND no interface for PLCI=0x101 CONNECT_IND ID=001 #0x040c LEN=0038 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = 8161 CallingPartyNumber = 09 8017 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default -- CONNECT_IND (PLCI=0x101,DID=61,CID=17,CIP=0x10,CONTROLLER=0x1) hfcpci: msn='*' DNID='61' MSN == hfcpci: Incoming call '17' - '61' -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk CAPI dial-in issues
It is connected to the PBX, alcatel omnipcx. My libcapi20is dated Oct 21, 2004. Where can I get the libcapi? There seems to be 100 sources and none smells official. On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote: Armin Schindler ha scritto: There are no more messages? SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not signal the call to Asterisk. The sending complete field is pretty new in the libcapi, maybe he just need to update the capi20 lib. Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)
Hello, I have the following setup: (*)---IP---Micronet 5012 H.323 box --- POTS --- PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing Dial(SIP/xlite1-7a03, H323/120/smallbox) in new stack --- h323_request - data 120/smallbox format 0x4 (ulaw) --- find_peer +++ find_peer +++ h323_request --- h323_call- 120/smallbox +++ h323_call -- Called 120/smallbox --- onNewCallCreated ooh323c_1 --- find_call +++ find_call Outgoing call smallbox(ooh323c_1) - Codec prefs - (gsm|alaw|ulaw) Adding capabilities to call(outgoing, ooh323c_1) Adding gsm capability to call(outgoing, ooh323c_1) Adding g711 alaw capability to call(outgoing, ooh323c_1) Adding g711 ulaw capability to call(outgoing, ooh323c_1) --- configure_local_rtp +++ configure_local_rtp +++ onNewCallCreated ooh323c_1 --- setup_rtp_connection --- find_call +++ find_call +++ setup_rtp_connection --- onAlerting ooh323c_1 --- find_call +++ find_call +++ onAlerting ooh323c_1 -- H323/smallbox-f14a is ringing --- onCallEstablished ooh323c_1 --- find_call +++ find_call +++ onCallEstablished ooh323c_1 -- H323/smallbox-f14a answered SIP/xlite1-7a03 -- Attempting native bridge of SIP/xlite1-7a03 and H323/smallbox-f14a --- h323_set_peer - H323/smallbox-f14a Sep 5 18:28:27 NOTICE[27211]: src/chan_h323.c:2749 h323_convertAsteriskCapToH323Cap: Don't know how to deal with mode 0x40 (slin) --- close_rtp_connection --- find_call +++ find_call +++ close_rtp_connection --- onCallCleared ooh323c_1 --- find_call +++ find_call --- h323_hangup hanging smallbox +++ h323_hangup == Spawn extension (default, 120, 1) exited non-zero on 'SIP/xlite1-7a03' --- h323_destroy Destroying smallbox +++ h323_destroy I think that, if it would not try to do native bridge, but transcode the sound, it would work. Perhaps there is an option, like forcetranscode? -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk CAPI dial-in issues
Oh, and I am using chan_cap via mISDN on HFCPCI. On 05/09/05, Konrads Smelkovs [EMAIL PROTECTED] wrote: It is connected to the PBX, alcatel omnipcx. My libcapi20is dated Oct 21, 2004. Where can I get the libcapi? There seems to be 100 sources and none smells official. On 05/09/05, Sergio Chersovani [EMAIL PROTECTED] wrote: Armin Schindler ha scritto: There are no more messages? SETUP or SENDING COMPLETE IE is missing and without it, chan_capi will not signal the call to Asterisk. The sending complete field is pretty new in the libcapi, maybe he just need to update the capi20 lib. Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Konrads Smelkovs Applied IT sorcery. -- Konrads Smelkovs Applied IT sorcery. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi hfcpci mISDN linux 2.6.12 not working
Hello,
These are error messages I get when I try to call a number over CAPI channel.
-- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack
-- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack
data = hfcpci/b17
capi request for interface 'hfcpci'
== hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00, ton=0x00)
-- hfcpci: received CONNECT_CONF PLCI = 0x201
-- Called hfcpci/b17
== hfcpci: Interface cleanup PLCI=0x201
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'SIP/xlite1-3b80' status is 'NOANSWER'
mISDNUser test tools show ISDN line working (testcon).
capi info shows that 2 B channels are available
capiinfo utility also dumps meaningful information - indicating that
it indeed recognises the card.
chan_capi version - CVS as of Sep 2, 2005 from chan-capi.sf.net
I read comments on voip-info about 2.6.12 kernel breaking something,
but the patch was for capi 0.3.5, not sure it applies...
capi.conf:
[hfcpci]
;;PointToPoint (55512-0)
isdnmode=ptp
incomingmsn=61
msn=61
controller=1
devices=2
context=capi-in
extensions.conf
[capi-in]
exten = 61,1,Dial,SIP/xlite1
exten = s,1,HangUp
[sip]
exten = _XX,1,SetCallerId(0)
exten = _XX,2,Dial(CAPI/hfcpci/b17)
capiinfo:
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller HFC1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
DTMF supported
Supplementary Services supported
B1 protocols support: 0x0003
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
ISO 7776 (X.75 SLP)
Transparent
Transparent (ignoring framing errors of B1 protocol)
B3 Transparent
ISO 8208 (X.25 DTE-DTE)
0100
0200
1800
0300
4300
0500
Supplementary services support: 0x0012
Terminal Portability
Call Forwarding
protocols support: 0x0005
--
Konrads Smelkovs
Applied IT sorcery.
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[Asterisk-Users] Working SIP phone for linux and windows
Hello I have yet to discover a software package that would both register and have ulaw codec. The SIP communicator (Java) came closest to usable, but didn't have the ulaw codec working. What do you use for communications? -- Konrads Smelkovs Applied IT sorcery. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN+w6692pci errors while loading
Hello It is all very confusing due to little information available :) I have a w6692 PCI card, so 1) What ports or modes i can use it? Currently i am plugged into a T0 port, can it be used? And what's the difference from S0? Please point me to some reading full of clues. 2) Due to lack of my understanding of the modes i can't seem to get the right protocol and layermask values for w6692pci.ko module at insmod time. There was this (http://lists.digium.com/pipermail/asterisk-users/2004-December/076239.html) discussion, but it is not helpful to me :( . Clues are very welcome. TIA. -- Konrads Smelkovs Applied IT sorcery. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More on W6692pci NT mode under chan_misdn
So far i've grasped that to use a card in NT mode it should have layermask=3 as module option. Is it the only thing that sets TE or NT mode for card? Perhaps there are settings in misdn.conf ? I can only get the card to work in TE mode and even then when asterisk is ran as asterisk -vvvgc it exits right after chan_misdn is loaded with theese messages: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 Locking Config Mutex UnLocking Config Mutex Init. Stack on port:1 TE Stack No lower Id port:1 init_stack: Success talkinghead:~ # syslog: Feb 18 16:58:50 talkinghead kernel: MISDN free_device: entitylist not empty in misdn.conf there is [NT cards] context=outgoing ports=1 with ports=1ptp it Segfaults. Clues? -- Konrads Smelkovs Applied IT sorcery. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
