Re: [asterisk-users] fax problem
Thank you francois!
Where could you find that info ?
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de F6HQZ
Envoyé : mercredi 23 décembre 2009 22:46
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] fax problem
Oops !
The sendmail macro was missing, sorry !
[macro-Sendmail]
;===
; ARG1 = Address To
; ARG2 = Address From
; ARG3 = File attachment
; ARG4 = Pages Qty
; ARG5 = Rate
; ARG6 = HeaderInfo
; ARG7 = RemoteID
; ARG8 = Resolution
;===
exten = s,1,NoOp( SENDMAIL )
exten = s,n,NoOp(To:${ARG1} From:${ARG2} Subject:Fax de ${ARG6}
Attach:${ARG3} Pg:${ARG4} Rate:${ARG5} HeaderInfo:${ARG6}
RemoteID:${ARG7} Res:${ARG8})
exten = s,n,System(echo Entete FAX : ${ARG6} - ${ARG4} pages -
Rate:${ARG5} - CID:${ARG7}, Resolution : ${ARG8}|/bin/mailx -s
FAX de : ${ARG6} - CID : ${ARG7} -a ${ARG3} -r ${ARG2} ${ARG1})
exten = s,n,NoOp( SENT )
exten = s,n,System(rm ${ARG3})
-Message d'origine-
De : F6HQZ [mailto:f6hq...@hamwlan.net]
Envoyé : mercredi 23 décembre 2009 22:44
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [asterisk-users] fax problem
Hi Francois,
here is Francois too ;-)
Check that :
[fax-outbound-calls]
exten = _X.,1,Dial(${ACROPOLIS}/${EXTEN},,G(fax-tx^send^1))
[fax-tx]
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Set(FaxTxDir=/var/spool/fax/tx/)
exten = send,n,Set(FAXFILEPDF=fax-${FAXCOUNT}-tx.pdf)
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten = send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = send,n,Set(FAXFILE=test.tif)
; Set FAXOPTs
exten = send,n,NoOp( SETTING FAXOPT )
exten = send,n,Set(FAXOPT(filename)=${FAXFILE})
exten = send,n,Set(FAXOPT(ecm)=yes)
exten = send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)}
at ${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten = send,n,Set(FAXOPT(localstationid)=0170619058)
exten = send,n,Set(FAXOPT(maxrate)=14400)
exten = send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = send,n,NoOp( SENDING FAX )
exten = send,n,SendFAX(${FaxTxDir}${FAXFILE}|d)
; Hangup! Print FAXOPTs
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
; Sendmail options for reports by email :
exten = h,n,System(/usr/bin/tiff2pdf -o ${FaxTxDir}${FAXFILEPDF}
${FaxRxDir}${FAXFILE})
exten =
h,n,macro(Sendmail,postmas...@acropolis.fr,aster...@acropolis.fr,${FaxRxDir}
${FAXFILEPDF},${FAXOPT(pages)},${FAXOPT(rate)},${FAXOPT(
headerinfo)},${FAXOPT(remotestationid)},${FAXOPT(resolution)})
Mainly extracted from the Digium FFA manual.
I hope this can help you.
Best Regards,
Francois
On Wed, Dec 23, 2009 at 10:49 AM, BERGANZ François
franc...@acropolistelecom.net wrote:
Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten = _X.,1,SendFax(/root/test.tiff)
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[asterisk-users] Tel uri Support
Hi All, Is someone implemented Tel uri support in the latest asterisk ? If yes, can you guys share some info on it Regards, Ramananda AS ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P clone card problem
Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf fxsks=1 echocanceller=mg2,1 loadzone= hu defaultzone = hu dahdi_tool OK Wildcard X100P Board 1 dahdi_tool (After select select) Current Alarms: No alarms ( Should I see OK here instead ? Is it normal ?) Sync Source:Internally clocked IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 1/ 1/ 1 1 TxA - TxB - TxC - TxD - RxA - RxB - RxC - RxD - dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 lsmod dahdi_echocan_mg2 9992 0 dahdi 193576 2 dahdi_echocan_mg2,wcfxo crc_ccitt 5888 1 dahdi wcfxo 14496 0 dahdi 193576 2 dahdi_echocan_mg2,wcfxo cat /proc/interrupts 18: 818521 IO-APIC-fasteoi wcfxo dmseg (after /etc/init.d/dahdi start) wcfxo :01:08.0: PCI INT A - Link[APC3] - GSI 18 (level, low) - IRQ 18 Found a Wildcard FXO: Wildcard X100P dahdi_echocan_mg2: Registered echo canceler 'MG2' dahdi: Registered tone zone 21 (Hungary) /etc/asterisk/chad_dahdi.conf [trunkgroups] [channels] language=nl rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes busycount=4 signalling=fxs_ks callgroup=1 pickgroup=1 group=1 context=pozivi channel = 1 /etc/asterisk/extensions.conf [agents] exten = 001,1,AgentLogin(1) exten = 001,2,Hangup [pozivi] exten = s,1,Answer exten = s,2,Queue(poziv|tTH) exten = s,3,Hangup exten = 2,1,Dial(dahdi/g1/0641842717) exten = 2,2,Hangup localhost*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault 1pozivi nl default localhost*CLI dahdi show channel 1 Channel: 1CLI File Descriptor: 20 Span: 1 Extension: Dialing: no Context: pozivi Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook localhost*CLI console dial 2 == Console is full duplex -- Executing [...@default:1] Dial(Console/dsp, dahdi/g1/0641842717) in new stack -- Called g1/0641842717 -- DAHDI/1-1 answered Console/dsp Console call has been answered ,but my mobile does not ring...Nothing happends. I checked wall phone socket with ordinary phone and I can get signal and call, but X100P plugged in at same socket gave me above result.. What's wrong ? Is it broken card or misconfiguration or removed support for X100P in dahdi ? Thank you, Uros -- Use Free Software http://www.fsf.org/ --- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't load cdr_radius.so module?
I'm not familiar with cdr_radius, but is there a debugging option? Anything in /var/log/messages? On Thu, Dec 24, 2009 at 1:35 AM, Zhang Shukun bit...@gmail.com wrote: Thank you ! i have load cdr_radius.so successfully! but another error occur. -- Executing [4...@tutorial:1] Dial(SIP/ivan-0a07dc80, SIP/test) in new stack -- Called test -- SIP/test-0a08b0f0 is ringing -- SIP/test-0a08b0f0 answered SIP/ivan-0a07dc80 -- Packet2Packet bridging SIP/ivan-0a07dc80 and SIP/test-0a08b0f0 [Dec 24 09:30:32] ERROR[10747]: cdr_radius.c:227 radius_log: Failed to record Radius CDR record! == Spawn extension (tutorial, 4321, 1) exited non-zero on 'SIP/ivan-0a07dc80' it says Failed to record Radius CDR record. Could you tell me , what's wrong with it? 2009/12/23 Olle E. Johansson o...@edvina.net: 23 dec 2009 kl. 11.25 skrev David Cunningham: Shukun, It tells you No such file or directory. Is the file in your modules directory? Actually, to be more specific. The module cdr_radius.so exists, but can't bind to the radius library libradiusclient-ng.so.2. Check LD_LIBRARY_PATH /O On Wed, Dec 23, 2009 at 10:09 AM, Zhang Shukun bit...@gmail.com wrote: hi , all when i do the command module load cdr_radius.so ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory [Dec 23 17:55:41] WARNING[31072]: loader.c:730 load_resource: Module 'cdr_radius.so' could not be loaded. -- Regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax problem
Ah ! It's a jamming of Digium FFA user manual, ideas and tests from my customers and myself. From Digium's side you can/must acces to this WEB page : https://www.digium.com/en/supportcenter/documentation/viewdocs/FAX I love to check Digium's solutions and to know how to use them. So, I have writed many how to's in french to install Asterisk and the main extra solutions, gateways, IP-Phones with provisionning, etc... I often send this little docs to my customers and friends, to win time and have a fast success. Best Regards, Francois ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with gdb
super quick asterisk in gdb howto: compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags) gdb asterisk run -cvv wait for the crash bt bt full and now make the patch :) Kristijan 2009/12/24 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Dec 24, 2009 at 12:13:55PM +0530, Goyal, Amit wrote: Hi All, Can some help me with how to run Asterisk with gdb. What specifically do you want to do? What do you want to check? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with gdb
On Thu, Dec 24, 2009 at 01:12:58PM +0100, Kristijan Vrban wrote: super quick asterisk in gdb howto: compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags) which changes the behaviour of your code. Rebuilding is not always an option. If using Asterisk from a package, be sure to install debug symbols. e.g. package asterisk-dbg or asterisk-debuginfo . gdb asterisk run -cvv wait for the crash or, grab a core file, and: gdb -c core.file /usr/sbin/asterisk bt bt full -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager API - DTMF issues
Hello users, i have been testing the DTMF tone detection using originate command both from Asterisk CLI and java API. but my DTMF entry at the originate user is not getting detected by the asterisk in both the cases what i should do to make it work any help will be appreciated. my versions i)asterisk 1.4.25 ii)SkyepeForAsterisk 1.4_1.0.6 asterisk CLI originate Skype/xx extension 1...@testing #extensions.conf [testing] exten = 1000,1,NoOp() exten = 1000,n,BackGround(welcome) exten = 1,1,Dial(SIP/101,20,r) exten = 1,n,Hangup() exten = 2,1,Dial(SIP/102,20,r) exten = 2,n,Hangup() Thanks in advance srinivas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create MeetME room with dialplan?
Whether you can do this and how successfully depends on your Asterisk Flavor. Meetme has no internal limitation that would allow you to limit the meetme to two callers. IMO you would have better luck using a bank of pre-defined meetme rooms, but you can set up as many as you want on the fly from the dialplan. The approach I would take to this is this Caller A calls in - he is assigned to pre-defined room x. AMI then calls caller B and dumps him/her into room x. You could use the Asterisk DB or MYSQL to track rooms by caller. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik Sent: Wednesday, December 23, 2009 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to create MeetME room with dialplan? Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both parties being called are done through a third party SIP provider. 1- So, I am looking to use 100s of calls to connect at with each two party connected to each other. I guess I need 50 meetme rooms for 100 callees?! 2- And, is it possible to create those meetme rooms on the go from within my context via dialplan? or is there a better approach that I should take? Thanks, Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Faxing
Barry Fawthrop wrote: Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in libc-2.7.so[b7e3f000+155000] asterisk[2647]: segfault at 30353466 ip b7e6338b sp bfb7708c error 4 in libc-2.7.so[b7ded000+155000] asterisk[2666]: segfault at 30353466 ip b7dfe38b sp bfa6e4ec error 4 in libc-2.7.so[b7ded000+155000] When I use dahdi at least asterisk will start but app_rxfax will fail with unknown symbol in ast_register_application I could only find a precompiled-linux-spandsp-app-fax is there anyway to get the source and compile for myself? Where I can compile for dahdi and not zaptel. Or can someone explain why the segfaulting?? my machine O/S is: Debian kernel : 2.6.26-2-686 i686 My goal is to connect a fax machine to the the sangoma card so I can send paper based faxes. I have a teliax provided SIP phone number which will be the fax number to receive all faxes and have them emailed to a central email address, hopefully in PDF format. where they can be printed and/or forwarded. It would be nice to have the incoming fax emailed to a specific address based on either subject or senders phone number. If this is possible I would like to know how. Thanks in advance Barry Looking at SIP debug of an incoming fax call I see v=0 o=root 2007366114 2007366115 IN IP4 24.xx.xx.xx s=Asterisk PBX 1.6.2.0 c=IN IP4 24.xx.xx.xx t=0 0 m=image 4308 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC Yes my udptl.conf file has udptlstart = 4000 udptlend = 4999 T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 VoipFaxMaxRate = 5 T38MaxBitRate = 14400 ;udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes ;; tried both no and yes It would appear from the BitRate and UdpEC that my udptl.conf file is not being used? Or am I missing something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow and language
Administrator TOOTAI a écrit : Hi, I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip extension definition, when I set language, it is not reported in the extensions_custom.conf file (eg language=xx). Am I missing something or is it not the right way to set language? Hello, sorry to insist on this, does nobody use AsteriskNow? I register to the AsteriskNow mailing list, no more luck to get answer. I also notice that call-limit was setted to 50! Where can I modify thos options. Thanks for any hint. Merry christmas -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Faxing
Barry Fawthrop escribió: Barry Fawthrop wrote: Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in libc-2.7.so[b7e3f000+155000] asterisk[2647]: segfault at 30353466 ip b7e6338b sp bfb7708c error 4 in libc-2.7.so[b7ded000+155000] asterisk[2666]: segfault at 30353466 ip b7dfe38b sp bfa6e4ec error 4 in libc-2.7.so[b7ded000+155000] When I use dahdi at least asterisk will start but app_rxfax will fail with unknown symbol in ast_register_application I could only find a precompiled-linux-spandsp-app-fax is there anyway to get the source and compile for myself? Where I can compile for dahdi and not zaptel. Or can someone explain why the segfaulting?? my machine O/S is: Debian kernel : 2.6.26-2-686 i686 My goal is to connect a fax machine to the the sangoma card so I can send paper based faxes. I have a teliax provided SIP phone number which will be the fax number to receive all faxes and have them emailed to a central email address, hopefully in PDF format. where they can be printed and/or forwarded. It would be nice to have the incoming fax emailed to a specific address based on either subject or senders phone number. If this is possible I would like to know how. Thanks in advance Barry Looking at SIP debug of an incoming fax call I see v=0 o=root 2007366114 2007366115 IN IP4 24.xx.xx.xx s=Asterisk PBX 1.6.2.0 c=IN IP4 24.xx.xx.xx t=0 0 m=image 4308 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC Yes my udptl.conf file has udptlstart = 4000 udptlend = 4999 T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 VoipFaxMaxRate = 5 T38MaxBitRate = 14400 ;udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes ;; tried both no and yes It would appear from the BitRate and UdpEC that my udptl.conf file is not being used? Or am I missing something? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please correct me if I'm wrong, but AFAIK spandsp based fax applications for asterisk only support a maximum of 9600bps. Cheers and Merry Christmas, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Faxing
Miguel Molina wrote: Please correct me if I'm wrong, but AFAIK spandsp based fax applications for asterisk only support a maximum of 9600bps. No. V.17 (speeds up to 14400 bps) are supported. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording the Calls to a USB Drive
Hi Guys, Merry Christmas and Happy new Year. I am looking for some assistance from the group as i think this might already have been tried before. i have an asterisk server with a external USB Harddisk Drive, just to store recordings. I am using the mixmonitor application for doing the recordings. When i have active calls that are being recorded to the USB Drive, and if my USB disk fails for some odd reason, like hardware failure or power failure..., asterisk complains that it is unable to write the recording to the USB Drive either by crashing asterisk or generate an infinite loop of errors on the asterisk console (Input/Ouptut Errors). If I try to unload the module app_mixmonitor.so, asterisk crashes. I am wondering if we can make asterisk stop recording on all the recorded calls and not to crash/generate errors if it does not see the USB drive any more. i thought the easiest way is to unload the app_mixmonitor module, but unfortunately it is crashing asterisk at the same time. Thanks Krishna ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check Asterisk SIP registration
Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording the Calls to a USB Drive
Just my opinion; unless you are recording long or many long calls, you should record to your local drive, then copy the files to the USB drive. Asterisk is a very good tool - you don't need to mess it up by introducing an easy point of failure. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Krishna Sumanth Chava Sent: Thursday, December 24, 2009 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recording the Calls to a USB Drive Hi Guys, Merry Christmas and Happy new Year. I am looking for some assistance from the group as i think this might already have been tried before. i have an asterisk server with a external USB Harddisk Drive, just to store recordings. I am using the mixmonitor application for doing the recordings. When i have active calls that are being recorded to the USB Drive, and if my USB disk fails for some odd reason, like hardware failure or power failure..., asterisk complains that it is unable to write the recording to the USB Drive either by crashing asterisk or generate an infinite loop of errors on the asterisk console (Input/Ouptut Errors). If I try to unload the module app_mixmonitor.so, asterisk crashes. I am wondering if we can make asterisk stop recording on all the recorded calls and not to crash/generate errors if it does not see the USB drive any more. i thought the easiest way is to unload the app_mixmonitor module, but unfortunately it is crashing asterisk at the same time. Thanks Krishna ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
It looks to me like calls from your Dial will route back to the sip-outgoing context and Dial again... it's loop. You'd really need to provide more logging information to advise further. On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi motamed...@gmail.com wrote: On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham dcunning...@voisonics.com wrote: AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might help you with the Host '192.168.0.139' does not implement 'REGISTER' problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi motamed...@gmail.comwrote: On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.comwrote: On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote: On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: Dear All I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf extensions.conf as the followings : Under sip.conf : - [general] register = toronto:welc...@192.168.0.139/osaka [osaka] type=friend secret=welcome context=osaka_incoming host=dynamic disallow=all allow=alaw [6672019] type=friend host=dynamic context=phones Try this: [general] register = toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance [general] ;register = toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I downloaded installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US
Re: [asterisk-users] how to check Asterisk SIP registration
sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check Asterisk SIP registration
Thanks but sip show registry yields nothing. --- On Thu, 12/24/09, Danny Nicholas da...@debsinc.com wrote: sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording the Calls to a USB Drive
On Thu, Dec 24, 2009 at 11:24:24AM -0500, Krishna Sumanth Chava wrote: Hi Guys, Merry Christmas and Happy new Year. I am looking for some assistance from the group as i think this might already have been tried before. i have an asterisk server with a external USB Harddisk Drive, just to store recordings. I am using the mixmonitor application for doing the recordings. When i have active calls that are being recorded to the USB Drive, and if my USB disk fails for some odd reason, like hardware failure or power failure..., asterisk complains that it is unable to write the recording to the USB Drive either by crashing asterisk or generate an infinite loop of errors on the asterisk console (Input/Ouptut Errors). If I try to unload the module app_mixmonitor.so, asterisk crashes. IIRC such a write to a disk will not be using DMA, and hence will take much more CPU. I suggest you do some load testing first to see if this is not an issue. I am wondering if we can make asterisk stop recording on all the recorded calls and not to crash/generate errors if it does not see the USB drive any more. i thought the easiest way is to unload the app_mixmonitor module, but unfortunately it is crashing asterisk at the same time. You get a udev even when a device is removed. Get that event to do whatever changes are required in the dialplan. That said, you will be losing data if you remove the device while a call is recorded. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording the Calls to a USB Drive
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote: Just my opinion; unless you are recording long or many long calls, you should record to your local drive, then copy the files to the USB drive. Asterisk is a very good tool - you don't need to mess it up by introducing an easy point of failure. Yes. I do this since 3 years and work very well. -- Best regards, Gergomailto:csi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create MeetME room with dialplan?
Hello, Thanks for the reply. I am in full control of the meetme rooms since I initiate the call for both parties and I can do next call into a new meetme room. Can anyone please share their AMI, PHP, and dialplan code relating to creating MeetME rooms on the go? Much appreciated. Thanks On Thu, Dec 24, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: Whether you can do this and how successfully depends on your Asterisk Flavor. Meetme has no internal limitation that would allow you to limit the meetme to two callers. IMO you would have better luck using a bank of pre-defined meetme rooms, but you can set up as many as you want “on the fly” from the dialplan. The approach I would take to this is this Caller A calls in – he is assigned to pre-defined room x. AMI then calls caller B and dumps him/her into room x. You could use the Asterisk DB or MYSQL to track rooms by caller. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce Nik *Sent:* Wednesday, December 23, 2009 10:53 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] How to create MeetME room with dialplan? Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both parties being called are done through a third party SIP provider. 1- So, I am looking to use 100s of calls to connect at with each two party connected to each other. I guess I need 50 meetme rooms for 100 callees?! 2- And, is it possible to create those meetme rooms on the go from within my context via dialplan? or is there a better approach that I should take? Thanks, Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Troubleshooting help
It looks like whatever is being transmitted, or the response, isn't getting through. Possibly due to NAT or a firewall? It would help if you described the scenario where this is occurring. On Thu, Dec 24, 2009 at 7:18 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:13] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:14] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:15] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:16] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:19] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:20] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:22] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:24] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:26] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:28] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:30] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:32] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaae...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:35] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording the Calls to a USB Drive
On Thu, 2009-12-24 at 18:53 +0100, Gergo Csibra wrote:
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you don't need to mess it up by introducing
an easy point of failure.
Yes. I do this since 3 years and work very well.
What would be the problem with mounting the usb disc somewhere like:
/mnt/usbdisk and using something like:
exten = s,2,MixMonitor(/mnt/usbdisc/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%
S)}-${UNIQUEID}.wav,v(0))
???
This should be good for anything capable of being mounted (smb, nfs et
al). That's one of the beautiful things about Linux. It does not care
what the device is - just that it can find it.
Of course, the caveat - if it's not mounted, it can't write - but I'm
sure the excellent developers of Asterisk have coded to catch basic
exceptions like 'file not found'.
___
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Re: [asterisk-users] 1.6 Troubleshooting help
Dave Wrote: It looks like whatever is being transmitted, or the response, isn't getting through. Possibly due to NAT or a firewall? It would help if you described the scenario where this is occurring. Indeed, my post was gibberish :-O This was a 'nat' issue, but not in the traditional sense. Draytek router getting it's knickers in a twist and not wanting to play happy sockets. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Core show function?
On Wednesday 23 December 2009 12:52:38 Ira wrote: Someone posted a message suggesting someone try sendtext() and so I thought I'd see if it was useful. Much searching through help at the CLI has failed to find any help for sendtext, but I did find that: core show function vmcount fails but: core show function VMCOUNT works. Is that a bug and if so, has it been reported? It's not a bug. Function names are case-sensitive. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check Asterisk SIP registration
I wrote a script to check clients and restart asterisk if registrations died (external IAX)...but you could modify for your needs. Check it out on www.generationd.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 12:06 PM To: Asterisk Users List Subject: Re: [asterisk-users] how to check Asterisk SIP registration Thanks but sip show registry yields nothing. --- On Thu, 12/24/09, Danny Nicholas da...@debsinc.com wrote: sip show registry might be more helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Thursday, December 24, 2009 10:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] how to check Asterisk SIP registration Unfortunately, sip show peers did not work in my case. The sip peers were apparently online and OK (I use qualify=yes) but they weren't... The SIP clients could NOT register, so they were offline but sip show peers stated that they were OK. I would prefer to perform an automated SIP registration (via cron script). If it fails then I can spawn a rescue script. Surely, a real sip registration is more reliable then sip show peers. Any ideas? Vieri --- On Wed, 12/23/09, Danny Nicholas da...@debsinc.com wrote: Sip show users or sip show peers should do the trick, but I'm not sure about 1.2; all of my experience is in the 1.4 branch. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, December 23, 2009 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to check Asterisk SIP registration Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a graceful restart). Besides, would a restart gracefully actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again solved the problem. How can I diagnose what happened to the SIP service (I checked the log but am quite lost)? Also, how can I do a simple command-line check to see that SIP registrations are OK? I would like to use a SIP client (like sipsak) to perform a simple registration from a custom bash script so I can quickly detect if this problem occurs again and auto-kill+restart the asterisk process. I know this sounds ugly but on my production server, it's better to bring the whole system down and back up in as little time as possible. Any suggestions? Asterisk is 1.2.31.1 Some log lines: Dec 23 13:13:16 DEBUG[11482] channel.c: Avoiding initial deadlock for 'SIP/4053-b4520e98' Dec 23 13:13:16 WARNING[11482] channel.c: Avoided initial deadlock for '0xb4302278', 9 retries! Dec 23 13:13:43 VERBOSE[18837] logger.c: -- Executing Dial(SIP/6174-b456d828, SIP/4062|20|tTwWM(auto-blkvm)) in new stack Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Dec 23 13:13:43 VERBOSE[18837] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Dec 23 13:13:43 DEBUG[18837] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
Hello,
Please forgive me if I'm repeating this post. I have searched and looked for
similar problem with a solution but have not see a similar one.
My outgoing SIP and other channels work fine but the incoming/inbound SIP
call goes straight to Broadvoice voicemail. I see that Broadvoice is
registered when I look at the SIP registry. I have turned on SIP Debug and
it is below.
Anyone know why even when SIP has registered I do not see incoming calls?
Thanks,
--extensions.conf
[global]
[general]
bindport=5060
bindaddr = 0.0.0.0
deny=0.0.0.0/0.0.0.0
externhost=xyz.dyndns.org
localnet = 192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
allow=gsm
delayreject=yes
nochecksums=no
allowguest=no
delayreject=yes
pedantic=no
register = 703xxxy...@sip.broadvoice.com:s
ecurepassword:703xxxy...@sip.broadvoice.com/5000
[5000]
type=friend
context=internal-phones
secret=xxx
qualify=yes
host=dynamic ; behind nat
dtmfmode=rfc2833
[5002]
type=friend
context=internal-phones
secret=test
qualify=yes
host=dynamic ; behind nat
nat=yes
dtmfmode=rfc2833
[enter_broadvoice]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=703XXX
secret=securepassword
username=703XXX
insecure=very
;insecure=port,invite
context=incoming
authname=703XXX
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
extensions.conf
[globals]
[general]
autofallthrough=yes
[incoming_calls]
exten = 1703XXX,1,Dial(SIP/5000)
[internal-phones]
include = outgoing
exten = 5000,1,Dial(SIP/5000,20)
exten = 5002,1,Dial(SIP/5002,20)
[outgoing]
exten = _X.,1,NoOp()
exten = _X.,n,Dial(SIP/enter_broadvoice/${EXTEN})
SIP Registry--
-*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
sip.broadvoice.com:5060N 703xxxy...@s23
Registered Fri, 25 Dec 2009 01:14:03
SIP Debug--
-*CLI
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.226:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8;rport
Max-Forwards: 70
From: sip:703xxxy...@sip.broadvoice.comsip%3a703xxxy...@sip.broadvoice.com
;tag=as376e46ae
To: sip:703xxxy...@sip.broadvoice.com sip%3a703xxxy...@sip.broadvoice.com
Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
CSeq: 147 REGISTER
User-Agent: Asterisk PBX 1.6.1.6
Expires: 120
Contact: sip:5...@68.100.65.3 sip%3a5...@68.100.65.3
Content-Length: 0
---
Really destroying SIP dialog '65a8d48738a00d121fc9050e4771d...@192.168.1.51'
Method: REGISTER
Suuban*CLI
--- SIP read from UDP://147.135.32.225:5060 ---
SIP/2.0 200 OK
Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
CSeq: 147 REGISTER
From: sip:703xxxy...@sip.broadvoice.comsip%3a703xxxy...@sip.broadvoice.com
;tag=as376e46ae
To: sip:703xxxy...@sip.broadvoice.com sip%3a703xxxy...@sip.broadvoice.com
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8
Contact: sip:5...@68.100.66.3 sip%3a5...@68.100.66.3
Expires: 30
Content-Length:0
-
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '
65a8d48738a00d121fc9050e4771d...@192.168.1.51' in 32000 ms (Method:
REGISTER)
[Dec 25 00:42:31] NOTICE[2898]: chan_sip.c:16740 handle_response_register:
Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling
reregistration in 23 s)
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.226:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655;rport
Max-Forwards: 70
From: sip:703xxxy...@sip.broadvoice.comsip%3a703xxxy...@sip.broadvoice.com
;tag=as54c6327a
To: sip:703xxxy...@sip.broadvoice.com sip%3a703xxxy...@sip.broadvoice.com
Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
CSeq: 148 REGISTER
User-Agent: Asterisk PBX 1.6.1.6
Expires: 120
Contact: sip:5...@68.100.66.3 sip%3a5...@68.100.66.3
Content-Length: 0
---
Really destroying SIP dialog '65a8d48738a00d121fc9050e4771d...@192.168.1.51'
Method: REGISTER
Suuban*CLI
--- SIP read from UDP://147.135.32.226:5060 ---
SIP/2.0 200 OK
Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
CSeq: 148 REGISTER
From: sip:703xxxy...@sip.broadvoice.comsip%3a703xxxy...@sip.broadvoice.com
;tag=as54c6327a
To: sip:703xxxy...@sip.broadvoice.com sip%3a703xxxy...@sip.broadvoice.com
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655
Contact: sip:5...@68.100.66.3 sip%3a5...@68.100.66.3
Expires: 30
Content-Length:0
--
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[asterisk-users] compile issues.
Hi all, I am new to Asterik. Ia m trying to compile the source with the latest asterisk-1.6.2.0 these are the issues I am getting. initially,I got mkdir: cannot create directory `/var/lib/asterisk' than after reading the archives: I did: ./configure --enable-dev-mode --prefix=/tmp/asterisk --sysconfdir=/tmp/astconf --localstatedir=/tmp/aststate and than make install.: This is the error I am getting: tar: vm-tocallnum.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tocancel.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tocancelmsg.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toenternumber.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toforward.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tohearenv.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tomakecall.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tooshort.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toreply.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-torerecord.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-undelete.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-undeleted.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-unknown-caller.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-whichbox.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-youhave.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: Error exit delayed from previous errors make[1]: *** [/tmp/aststate/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.16] Error 2 make[1]: Leaving directory `/home/qchat/asterisk-1.6.2.0/sounds' make: *** [datafiles] Error 2 can any one help me to fix this issue. looks to be a basic thing I am missing ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
- Qurba Joog qurbaj...@gmail.com wrote:
| Hello,
|
| Please forgive me if I'm repeating this post. I have searched and looked for
similar problem with a solution but have not see a similar one.
|
| My outgoing SIP and other channels work fine but the incoming/inbound SIP
call goes straight to Broadvoice voicemail. I see that Broadvoice is registered
when I look at the SIP registry. I have turned on SIP Debug and it is below.
|
| Anyone know why even when SIP has registered I do not see incoming calls?
|
| Thanks,
|
|
| --extensions.conf
| [global]
|
| [general]
| bindport=5060
| bindaddr = 0.0.0.0
| deny= MailScanner warning: numerical links are often malicious:
0.0.0.0/0.0.0.0
| externhost= xyz.dyndns.org
| localnet = MailScanner warning: numerical links are often malicious:
192.168.1.0/255.255.255.0
| disallow=all
| allow=ulaw
| allow=gsm
| delayreject=yes
| nochecksums=no
| allowguest=no
| delayreject=yes
| pedantic=no
|
| register = 703xxxy...@sip.broadvoice.com:s
ecurepassword:703xxxy...@sip.broadvoice.com/5000
|
| [5000]
| type=friend
| context=internal-phones
| secret=xxx
| qualify=yes
| host=dynamic ; behind nat
| dtmfmode=rfc2833
|
| [5002]
| type=friend
| context=internal-phones
| secret=test
| qualify=yes
| host=dynamic ; behind nat
| nat=yes
| dtmfmode=rfc2833
|
| [enter_broadvoice]
| type=peer
| user=phone
| host= sip.broadvoice.com
| fromdomain= sip.broadvoice.com
| fromuser=703XXX
| secret=securepassword
| username=703XXX
| insecure=very
| ;insecure=port,invite
| context=incoming
| authname=703XXX
| dtmfmode=inband
| dtmf=inband
| ;Disable canreinvite if you are behind a NAT
|
| canreinvite=no
|
| extensions.conf
|
| [globals]
|
| [general]
|
| autofallthrough=yes
|
|
| [incoming_calls]
|
| exten = 1703XXX,1,Dial(SIP/5000)
|
| [internal-phones]
|
| include = outgoing
| exten = 5000,1,Dial(SIP/5000,20)
| exten = 5002,1,Dial(SIP/5002,20)
|
|
| [outgoing]
|
| exten = _X.,1,NoOp()
| exten = _X.,n,Dial(SIP/enter_broadvoice/${EXTEN})
|
| SIP Registry--
| -*CLI sip show registry
| Host dnsmgr Username Refresh State Reg.Time
| sip.broadvoice.com:5060 N 703xxxy...@s 23 Registered Fri, 25 Dec 2009
01:14:03
|
| SIP Debug--
| -*CLI
| REGISTER 11 headers, 0 lines
| Reliably Transmitting (no NAT) to MailScanner warning: numerical links are
often malicious: 147.135.32.226:5060 :
| REGISTER sip: sip.broadvoice.com SIP/2.0
| Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8;rport
| Max-Forwards: 70
| From: sip:703xxxy...@sip.broadvoice.com ;tag=as376e46ae
| To: sip:703xxxy...@sip.broadvoice.com
| Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
| CSeq: 147 REGISTER
| User-Agent: Asterisk PBX 1.6.1.6
| Expires: 120
| Contact: sip:5...@68.100.65.3
| Content-Length: 0
|
|
| ---
| Really destroying SIP dialog ' 65a8d48738a00d121fc9050e4771d...@192.168.1.51
' Method: REGISTER
| Suuban*CLI
| --- SIP read from UDP:// MailScanner warning: numerical links are often
malicious: 147.135.32.225:5060 ---
| SIP/2.0 200 OK
| Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
| CSeq: 147 REGISTER
| From: sip:703xxxy...@sip.broadvoice.com ;tag=as376e46ae
| To: sip:703xxxy...@sip.broadvoice.com
| Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8
| Contact: sip:5...@68.100.66.3
| Expires: 30
| Content-Length: 0
|
|
| -
| --- (9 headers 0 lines) ---
| Scheduling destruction of SIP dialog '
65a8d48738a00d121fc9050e4771d...@192.168.1.51 ' in 32000 ms (Method: REGISTER)
| [Dec 25 00:42:31] NOTICE[2898]: chan_sip.c:16740 handle_response_register:
Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling
reregistration in 23 s)
|
|
| REGISTER 11 headers, 0 lines
| Reliably Transmitting (no NAT) to MailScanner warning: numerical links are
often malicious: 147.135.32.226:5060 :
| REGISTER sip: sip.broadvoice.com SIP/2.0
| Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655;rport
| Max-Forwards: 70
| From: sip:703xxxy...@sip.broadvoice.com ;tag=as54c6327a
| To: sip:703xxxy...@sip.broadvoice.com
| Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
| CSeq: 148 REGISTER
| User-Agent: Asterisk PBX 1.6.1.6
| Expires: 120
| Contact: sip:5...@68.100.66.3
| Content-Length: 0
| ---
| Really destroying SIP dialog ' 65a8d48738a00d121fc9050e4771d...@192.168.1.51
' Method: REGISTER
| Suuban*CLI
| --- SIP read from UDP:// MailScanner warning: numerical links are often
malicious: 147.135.32.226:5060 ---
| SIP/2.0 200 OK
| Call-ID: 65a8d48738a00d121fc9050e4771d...@192.168.1.51
| CSeq: 148 REGISTER
| From: sip:703xxxy...@sip.broadvoice.com ;tag=as54c6327a
| To: sip:703xxxy...@sip.broadvoice.com
| Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655
| Contact: sip:5...@68.100.66.3
| Expires: 30
Re: [asterisk-users] compile issues.
On Thu, Dec 24, 2009 at 10:50:01PM -0800, Aditya Kumar wrote: Hi all, I am new to Asterik. Ia m trying to compile the source with the latest asterisk-1.6.2.0 these are the issues I am getting. initially,I got mkdir: cannot create directory `/var/lib/asterisk' than after reading the archives: I did: ./configure --enable-dev-mode --prefix=/tmp/asterisk --sysconfdir=/tmp/astconf --localstatedir=/tmp/aststate Those parameters don't really make sense, btw. /tmp tends to be periodaclly deleted. and than make install.: This is the error I am getting: tar: vm-tocallnum.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted Most likely something is not writable for your user. Were those files previously installed there by root? tar: vm-tocancel.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tocancelmsg.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toenternumber.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toforward.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tohearenv.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tomakecall.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-tooshort.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-toreply.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-torerecord.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-undelete.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-undeleted.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-unknown-caller.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-whichbox.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: vm-youhave.gsm: Cannot change ownership to uid 1000, gid 1000: Operation not permitted tar: Error exit delayed from previous errors make[1]: *** [/tmp/aststate/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.16] Error 2 make[1]: Leaving directory `/home/qchat/asterisk-1.6.2.0/sounds' make: *** [datafiles] Error 2 can any one help me to fix this issue. looks to be a basic thing I am missing -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
