[Asterisk-Users] simple AGI script
hello, does anybody have some agi script that can do following : when extension didn't pickup phone call, system send mail notify ( via sendmail ) to user mailbox with date, time and caller id ? ( something like missed call ) please can somebody help me with whis ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk -BEGIN GEEK CODE BLOCK- Version: 3.1 GS/E/B/PA/SS d+(++) s+:+ a C++$ ULS !P+++(---)$ L$ E++ W++ !N w(+++) !O() M++ V--() Y+$ PGP+ t- !5? X- !R !tv at b++() DI++ D+++ at G e+++ h(*) at r% --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] swissvoice ip10s
hallo, i would like to ask if somebody have sip firmware for this ip phone from swissvoice. they announced sip firmware in April 2004 but so far i'm unable to contact product manager and get the sip firmware. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom gsm codec
does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call Transfert with SwissVoice IP10S in MGCP mode
Daniel ANDRE wrote: I have the MGCP Firmware and call transfer doesn't work in my configuration. this is my mgcp.conf with working call transfer: [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfert with SwissVoice IP10S in MGCP mode
Daniel ANDRE wrote: Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the call manager. In my case, I have the message 'No available service' instead. What's wrong? Daniel call transfer in ip10s is possible only with mgcp formware... my phones works with h323... so no way... read notes from support : For the moment, call transfer is not yet fully integrated, so not proposed through the man machine interface. Call transfer will be ma naged through H.450. -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
rnc Info Lists wrote: Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. in sending you my mgcp.conf file, my ip10s mostly working fine... regards Marian ---mgcp.conf- [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi incoming call
Hello, i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line BRI, setup to point-to-multipoint. working fine, but when both channel on interfaces are used and somebody wants to call in - the response is ha-la-li - sounds like msn or called number does not exists. i think that right response should be - line busy tone... right ? can anybody tell me what is wrong ? regards Marian this apear in log : Oct 22 09:22:36 NOTICE[49156]: File chan_capi.c, Line 1813 (capi_handle_msg): CONNECT_IND ID=001 #0x4f00 LEN=0053 Controller/PLCI/NCCI= 0x301 CIPValue= 0x1 CalledPartyNumber = c15430754 CallingPartyNumber = 01 830903517715 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo BChannelinformation= 02 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default Oct 22 09:22:36 ERROR[49156]: File chan_capi.c, Line 1832 (capi_handle_msg): received a call waiting CONNECT_IND Oct 22 09:22:36 ERROR[49156]: File chan_capi.c, Line 1930 (capi_handle_msg): did not find device for msn = 5430754 Oct 22 09:22:36 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x4f01 Oct 22 09:22:36 NOTICE[49156]: File chan_capi.c, Line 1211 (pipe_msg): INFO_IND ID=001 #0x4f01 LEN=0023 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x70 InfoElement = c15430754 Oct 22 09:22:36 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x4f02 Oct 22 09:22:36 NOTICE[49156]: File chan_capi.c, Line 1211 (pipe_msg): INFO_IND ID=001 #0x4f02 LEN=0016 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x18 InfoElement = 80 Oct 22 09:22:36 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x4f03 Oct 22 09:22:40 NOTICE[49156]: File chan_capi.c, Line 1813 (capi_handle_msg): CONNECT_IND ID=001 #0x5097 LEN=0053 Controller/PLCI/NCCI= 0x301 CIPValue= 0x1 CalledPartyNumber = c15430754 CallingPartyNumber = 01 830903517715 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo BChannelinformation= 02 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line 1832 (capi_handle_msg): received a call waiting CONNECT_IND Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line 1930 (capi_handle_msg): did not find device for msn = 5430754 Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x5098 Oct 22 09:22:40 NOTICE[49156]: File chan_capi.c, Line 1211 (pipe_msg): INFO_IND ID=001 #0x5098 LEN=0023 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x70 InfoElement = c15430754 Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x5099 Oct 22 09:22:40 NOTICE[49156]: File chan_capi.c, Line 1211 (pipe_msg): INFO_IND ID=001 #0x5099 LEN=0016 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x18 InfoElement = 80 Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x509a Oct 22 09:22:58 DEBUG[901149]: File channel.c, Line 2245 (ast_channel_bridge): Didn't get a frame from channel: H323:6083 Oct 22 09:22:58 DEBUG[901149]: File channel.c, Line 2313 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/5427980]/9 and H323:6083 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 fritz-cards pci
hello, got anybody succesfully setup asterisk with three avm fritz pci cards - using the howto described in http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO i already have asterisk working with 2 cards, by when i add third card and compile driver ( see capiinit debug below ) asterisk freeze on capi initialization does anybody know how to solve this ? regards Marian -- Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capiutil.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/kernelcapi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capiutil.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/kernelcapi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capifs.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/fcpci.o Warning: loading fcpci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/f2pci.o Warning: loading f2pci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/f3pci.o Warning: loading f3pci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules fcpci - -(0)- - - - f2pci - -(0)- - - - f3pci - -(0)- - - - 1 fcpci running fritz-pciA1 3.10-02 0xdc00 5 2 f2pci running fritz2-pci A1 3.10-02 0xe000 12 3 f3pci running fritz3-pci A1 3.10-02 0xe400 10 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 call segmentation fault
hello, i have problem with oh323 channel driver (tryied differnet versions). when i try to make call between oh323 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial(H323:31119, SIP/92) in new stack -- Called 92 -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b answered H323:31119 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. 0:58.180 H245:8128d60 RTP_UDP No mediaControlChannel specified PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. Segmentation fault regads Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't use 2 controllers
Da t, 2003-09-04 at 11:40, Jamie Neil napsal: Simone Vasoli (BK s.r.l.) wrote: Hi, when I make a call, chan_capi always uses controller 2, and never uses controller 1 (so I have 4 lines for incoming calls, but only 2 lines instead of 4 for outgoing calls). this is with 2 AVM Fritz cards PCI. You can only use one Fritz PCI card per box due to a limitation built into the AVM CAPI driver. check this link - is describe how to use more than one fcpci in pc. I personally use 3 cards im my computers, without problems... http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO However you can get around this by hacking the (binary) fcpci kernel module and changing the name of second and subsequent modules. I haven't tried it personally, but I think there are several people who are using this technique sucessfully :). Just in case anyone is interested, AVM also say that you can't use the Fritz card and the B1 card in the same box. However I have found it seems to work fine provided the B1 CAPI driver is loaded *after* the Fritz driver. -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] avm fritz pci
hello, does anybody know how to setup avm fritz pci card in p2p mode ? regards marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi ptp mode
hello, i change my isdn line to ptp mode and i have some trouble to configure chan_capi in that way. What is the correct setup - e.g. if numbers are 123456x - where x can be from 0 to 9 ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi error
hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[393234]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. ERROR[311314]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1! will continue searching. NOTICE[311314]: File chan_capi.c, Line 1060 (capi_request): didn't find capi device with outgoing msn = 5430754. you should check your config! NOTICE[311314]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'CAPI' does anybody know how to solve this ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] caller id
Da St, 2003-07-09 at 18:26, Tan Aks napsal: Use SetCallerID(1234567). still not working.. after setting SetCallerID(1234) it displaying [EMAIL PROTECTED] is it just problem with snom phones ? or am i missing something ? regards Marian Tan telappliant.com - Original Message - From: Marian Danisek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 09, 2003 3:23 PM Subject: [Asterisk-Users] caller id Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connect 2 asterisk boxes
Hello, can anyone help me ho to simply connect 2 asterisk boxes, which are in different loactions e.g. with iax protocol ? some simple config files.. regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dbget dbput
Hi, do i need some other software than asterisk to use database commands - dbput and dbget in asterisk ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller id
Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and uclinux
Hello,every one! I would like to know if asterisk could run under uclinux. look at the archives in jun-2003, we spoke about how to compile in under uclibc - there is a patch. but i personaly had problem to run it, because of i cannot run asterisk a as a daemon ( fork ).. i thing beacuse uClinux can only do vfork(). cheers Marian Regards. -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seg Fault!!
We had the same problem, we fixed it downgrading the Capi version to 0.2.1b what's the diff between 0.2.1b and 0.2.2 ? regards Marian Salut, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de WipeOut . Enviado el: mircoles, 02 de julio de 2003 14:22 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Seg Fault!! Hi, I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am running chan_capi 0.2.2.. When a call is received Asterisk seg faults.. Not sure what information would be usefull to post so let me know what info will help to debug the problem.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asteri sk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compile in uclibc enviroment
hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian - gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o -ldl -lpthread -lncurses -lm -lresolv editline/libedit.a db1-ast/libdb1.a enum.o: In function `ast_get_enum': /usr/src/asterisk-cvs/enum.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/enum.c:307: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/enum.c:325: undefined reference to `__res_nclose' enum.o: In function `parse_naptr': /usr/src/asterisk-cvs/enum.c:157: undefined reference to `__dn_expand' srv.o: In function `ast_get_srv': /usr/src/asterisk-cvs/srv.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/srv.c:282: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/srv.c:297: undefined reference to `__res_nclose' srv.o: In function `parse_srv': /usr/src/asterisk-cvs/srv.c:136: undefined reference to `__dn_expand' collect2: ld returned 1 exit status make: *** [asterisk] Error 1 -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compile in uclibc enviroment
Hi, Here it is, attached. Adds a setting in the Makefile where enum support can be turned off. There will probably be some offset when patching due to other changes in my sources. Steve now * compile without errors... but to start * i made following entry in modules.conf under [modules] section : noload = app_enumlookup.so regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uclibc enviroment #2
ok i have another problem - howto run asterisk as a daemon ( fork ) in uclibc enviroment ? uClinux can only do vfork() a i think this is problem... does anybody know how to solve this ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom phones and redirect
Hello, does anybody suceesfully setup snom phones with sip firmware with asterisk to redirect call when phone is set to redirect if busy/ or allways redirect ? My console says : chan_sip.c Line 3000 (handle response) Dunno anything about a 302 Moved Temporarily from SIP... regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
