Re: [asterisk-users] CID not working.
Hi, Where is that file? I am using Asterisknow 1.5. Please tell me the location of the file * Thanks, Arun S* On Wed, Dec 30, 2009 at 8:35 PM, Danny Nicholas da...@debsinc.com wrote: How is DAHDI-1 set up in users.conf? You need something like this ; Span 2: WCTDM/4 Wildcard TDM400P REV I Board 5 [4001] fullname = Line 1 cid_number = 5551212 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Arun Sasidhar *Sent:* Wednesday, December 30, 2009 8:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.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] CID not working.
Hi, It is not working. The same error and no CID is the result. Thanks, Arun S On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC anth...@handynetworks.com wrote: You need to wait at least 1 second on an incoming POTS line for CID info, add a wait(1) as the first step on incoming connections. Thank you and have a nice day, Anthony Francis *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Arun Sasidhar *Sent:* Wednesday, December 30, 2009 7:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack *My chan_dahdi.conf file is as like this.* vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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 -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.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
[asterisk-users] Asterisk recieves 11 when pressing 1 from SIPphone
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing 1 from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my 1 as 11 ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) via SIP INFO [Dec 31 10:45:40] WARNING[17928]: pbx.c:2518 __ast_pbx_run: Invalid extension '33', but no rule 'i' in context ...[snip]... Same problem when pressing 3... Thank you. Jonas. ___ -- 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 recieves 11 when pressing 1 from SIPphone
jonas kellens wrote: [Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing 1 from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my 1 as 11 ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) via SIP INFO [Dec 31 10:45:40] WARNING[17928]: pbx.c:2518 __ast_pbx_run: Invalid extension '33', but no rule 'i' in context ...[snip]... Same problem when pressing 3... Thank you. Jonas. It may be me, but it looks like Asterisk correctly interprets the information, as the phone is configured to send both via RTP (once) and SIP INFO (twice). Your config tells the phone to send the digits twice, so Asterisk sees them twice... 1 twice makes 11, 3 twice makes 33! Try changing the phone's config to only use either RTP *or* SIP INFO... Good luck! --FP ___ -- 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 recieves 11 when pressing 1 from SIPphone
Francesco, Marking only RTP or only SIP info makes my DTMF to be correctly received by Asterisk (read: only once). It works fine now. Thanks. Jonas. On Thu, 2009-12-31 at 10:53 +0100, Francesco Peeters wrote: Jonas. It may be me, but it looks like Asterisk correctly interprets the information, as the phone is configured to send both via RTP (once) and SIP INFO (twice). Your config tells the phone to send the digits twice, so Asterisk sees them twice... 1 twice makes 11, 3 twice makes 33! Try changing the phone's config to only use either RTP *or* SIP INFO... ___ -- 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:Asterisk different codec schemes?
hadi motamedi wrote: Can you please let me know if we can have different codec schemes for audio codec in audio codec out ? I mean , in one application , we can have our audio codec input set to G.711 a-law and our audio codec output set to G.711 u-law . I am facing with an application that calls for such a settings . Asterisk does not support asymmetric codec configurations. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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] T.38 and Linksys SPA8000
2009/12/29 Vinícius Fontes vinic...@canall.com.br: Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. So you're trying to use a SPA8000 to act as a gateway, then you're trying to put asterisk somewhere. And there's a real fax machine involved. I don't think you've given enough information for anybody to actually tell you what went wrong with your 'transmission error'. So here are my suggestions: * if you're not already using the latest 1.6, you should be. A lot of T.38 and faxing-in-general improvements have been released. * are you sure the T.38 is actually negotiating? * if you leave T.38 disabled, everything should try to pass traditional audio over the VoIP codec of your choosing, and if you have a fairly reliable network, you MAY be able to accomplish faxing in that manner. * And now for my personal opinion... if you need reliable faxing, you shouldn't be using a SIP trunk into the premise, but rather a real traditional phone line. Fax is a hack to play tones to represent an image. VoIP is a hack to try to reduce the quality of a voice call without a human noticing. Put those together over the internet, and you shouldn't expect reliable faxing. Fax machines are much less forgiving than a human. If you lose data in the middle of a fax, the fax machine gets confused and can drop a fax that was mostly received okay. * I do think fax over voip has good usages. That means if you have a dedicated line to your voip provider with reliable quality, you would probably be okay. If you have a dedicated LAN that you are using internally for voip fax you are also probably okay. * if this whole project is an effort to save money and avoid the Brazil telco monopoly, you should consider paying for one of those services that allows you to send and receive faxes using a third-party, and not try to receive faxes at the premise. ___ -- 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] Friday Jan 1 Voip Users Conference
Thanks to Digium, the company, and to all of the fine people from Digium who participate in the weekly VoIP Users Conference conference! We will be live on Friday January 1, 2010 and there is also a reel of recorded greetings from people around the world wishing the VoIP Community a Happy New Year. You can hear this anytime during the year by downloading it from the site starting next week. Besides hangover remedies, live participants will be talking about the decade in VoIP and maybe what's to come. January's schedule has Tim Behrsin from Voxbone on iNum on January 8th, Hacking VoIP author Himanshu Dwivedi on January 15th and a guest from Plantronics on January 29th. Sometime in the coming weeks, Markus Feilner, author of “Beginning OpenVPN 2.0.9“. will be with us. When we have authors, their publishers usually give us a couple of books to give away as well. Until then, I wish all of you in this community the best of all possible combination of health, happiness, prosperity and minimal jitter. /r http://vuc.me Call (518) VUC-VOIP and say Happy New Year ___ -- 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] identifying channel for softhangup
Hi, I'm issuing a Bridgeaction through the manager interface. One Person is called, when answered second one is called first gets MoH. After the second person answers both channels are bridged together. Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week)) asterisk crashes. I suspected res_musiconhold and updated to the latest Version (repository) but nothing changed. here are some backtraces( number 'd). I can offer various core dumps, dialplan etc: Help would be greatly appreciated as I don't get any further on this problem and I have no Idea what to do. Verision: Asterisk 1.6.0.19 03.dec #0 0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6 #1 0xb7d20ac4 in _IO_free_backup_area () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6 #6 0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at format_wav.c:363 #7 0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at file.c:697 #8 0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718 #9 0xb760088a in spawn_mp3 (class=0xb54f7a34) at res_musiconhold.c:501 #10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511 #11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90, f=0xb63b6c40) at channel.c:2514 #12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at channel.c:3001 #13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037 #14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90, ms=9967, cond=0, data=0x0) at channel.c:1297 #15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at channel.c:1309 #16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at res_musiconhold.c:905 #17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960, data=0xb54ff0b8) at pbx.c:951 #18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0, context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3, label=0x0, callerid=0xb63e7b70 49711XXX, action=E_SPAWN, found=0xb5501208, combined_find_spawn=1) at pbx.c:3138 #19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90, context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3, callerid=0xb63e7b70 49711XXX, found=0xb5501208, combined_find_spawn=1) at pbx.c:3605 #20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at pbx.c:3692 #21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965 #22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861 #23 0xb7c9c4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6 08. Dec #0 0xb8000424 in __kernel_vsyscall () #1 0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7d6d57d in _IO_file_seekoff () from /lib/tls/i686/cmov/libc.so.6 #6 0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6 #7 0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6 #8 0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at format_wav.c:352 #9 0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at file.c:697 #10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718 #11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at res_musiconhold.c:504 #12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511 #13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8, f=0xb650b810) at channel.c:2514 #14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at channel.c:3001 #15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037 #16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8, ms=1887, cond=0, data=0x0) at channel.c:1297 #17 0x080977a2 in ast_safe_sleep (chan=0xb650a2d8, ms=1) at channel.c:1309 #18 0xb765214c in moh_alloc (chan=0xb650a2d8, params=0xb6ade0b8) at res_musiconhold.c:905 #19 0x08107446 in pbx_exec (c=0xb650a2d8, app=0xb7a16d28, data=0xb6ade0b8) at pbx.c:951 #20 0x0810ee3f in pbx_extension_helper (c=0xb650a2d8, con=0x0, context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3, label=0x0, callerid=0xb6509a58 49711XXX, action=E_SPAWN, found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3138 #21 0x08110b76 in ast_spawn_extension (c=0xb650a2d8, context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3, callerid=0xb6509a58 49711XXX, found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3605 #22 0x081112ed in __ast_pbx_run (c=0xb650a2d8, args=0x0) at pbx.c:3692 #23 0x08112714 in pbx_thread (data=0xb650a2d8) at pbx.c:3965 #24 0x0816a6ed in dummy_start (data=0xb6502e98) at utils.c:861 #25 0xb7ceb4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #26 0xb7de649e in clone () from
[asterisk-users] Dialplans Holiday Dates
I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far. Does anyone have a suggestion as to how to approach this? I'm running Asterisk 1.4.2. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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] Dialplans Holiday Dates
Perhaps make the dates a database entry? The fixed dates would stay the same each year and you would adjust only the floating dates. Or, there are really few holidays in the year. (Unless you are a government or a bank) Simple intercept code in the dialplan would handle most businesses. Just write 5-10 cloned lines of date traps in the code to pass the calls or send them to a closed handler. That is less disk/system intensive that doing a disk access, except they would likely be in cache anyway. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham Sent: Thursday, December 31, 2009 9:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dialplans Holiday Dates I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far. Does anyone have a suggestion as to how to approach this? I'm running Asterisk 1.4.2. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- 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] Dialplans Holiday Dates
Myles Wakeham wrote:
I suspect others have done this sort of thing with Asterisk before, but
I've not found any resources so far.
exten =
317xxx,1,Gosub(holiday_check,s,1)
[holiday_check]
;
;* Break out current 2 digit month
;
exten = s,1,Gosub(todays_date,s,1)
;*
;* Look for database entry for match against
;* month and day. Store sound file name
;* to GREETING variable
;*
exten = s,n,MYSQL(Connect connid localhost anonymous '' holidays)
exten = s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
exten = s,n,MYSQL(Query resultid ${connid} SELECT greeting FROM
schedule WHERE month = ${MONTH} AND day = ${DAY})
exten = s,n,MYSQL(Fetch fetchid ${resultid} GREETING)
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,MYSQL(Clear ${resultid})
;***
;* If GREETING *BLANK, must be a holiday
;* jump to s,10. Else return from subroutine
;***
exten = s,n,GotoIf($["${GREETING}" != ""]?9:13)
;
;* Play Holiday message and return
;* from subroutine
;
exten = s,n,Wait(2)
exten = s,n,Playback(local/holidays/greet_begin)
exten = s,n,Playback(local/holidays/${GREETING})
exten = s,n,Set(_Holiday=YES)
exten = s,n,Return
[todays_date]
;
;* Break out current 2 digit hour
;
exten = s,1,Set(HOUR=${STRFTIME(${EPOCH},,%H)})
;
;* Break out current 2 digit day
;
exten = s,n,Set(DAY=${STRFTIME(${EPOCH},,%d)})
;
;* Break out current 2 digit month
;
exten = s,n,Set(MONTH=${STRFTIME(${EPOCH},,%m)})
;
;* Break out current 4 digit year
;
exten = s,n,Set(YEAR=${STRFTIME(${EPOCH},,%Y)})
;
;* Set TODAY to DAY/MONTH/YEAR
;
exten = s,n,Set(TODAY=${MONTH}/${DAY}/${YEAR})
exten = s,n,Return()
___
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[asterisk-users] Random crashes on Bridgeaction
Sorry wrong topic... Hi, I'm issuing a Bridgeaction through the manager interface. One Person is called, when answered second one is called first gets MoH. After the second person answers both channels are bridged together. Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week)) asterisk crashes. I suspected res_musiconhold and updated to the latest Version (repository) but nothing changed. here are some backtraces( number 'd). I can offer various core dumps, dialplan etc: Help would be greatly appreciated as I don't get any further on this problem and I have no Idea what to do. Verision: Asterisk 1.6.0.19 03.dec #0 0xb7d24572 in free () from /lib/tls/i686/cmov/libc.so.6 #1 0xb7d20ac4 in _IO_free_backup_area () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d213c9 in __underflow () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7d1d868 in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7d1fcf8 in _IO_sgetn () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7d12fe0 in fread () from /lib/tls/i686/cmov/libc.so.6 #6 0xb5e6fced in wav_read (s=0x87c7c08, whennext=0xb54fca14) at format_wav.c:363 #7 0x080dcebb in read_frame (s=0x87c7c08, whennext=0xb54fca14) at file.c:697 #8 0x080dcf48 in ast_readframe (s=0x87c7c08) at file.c:718 #9 0xb760088a in spawn_mp3 (class=0xb54f7a34) at res_musiconhold.c:501 #10 0xb7600909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511 #11 0x0809abc0 in ast_read_generator_actions (chan=0xb63b7a90, f=0xb63b6c40) at channel.c:2514 #12 0x0809c9e3 in __ast_read (chan=0xb63b7a90, dropaudio=0) at channel.c:3001 #13 0x0809cd50 in ast_read (chan=0xb63b7a90) at channel.c:3037 #14 0x08097743 in ast_safe_sleep_conditional (chan=0xb63b7a90, ms=9967, cond=0, data=0x0) at channel.c:1297 #15 0x080977a2 in ast_safe_sleep (chan=0xb63b7a90, ms=1) at channel.c:1309 #16 0xb760214c in moh_alloc (chan=0xb63b7a90, params=0xb54ff0b8) at res_musiconhold.c:905 #17 0x08107446 in pbx_exec (c=0xb63b7a90, app=0xb7a09960, data=0xb54ff0b8) at pbx.c:951 #18 0x0810ee3f in pbx_extension_helper (c=0xb63b7a90, con=0x0, context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3, label=0x0, callerid=0xb63e7b70 49711XXX, action=E_SPAWN, found=0xb5501208, combined_find_spawn=1) at pbx.c:3138 #19 0x08110b76 in ast_spawn_extension (c=0xb63b7a90, context=0xb63b7cd8 Click2Call4_0, exten=0xb63b7d28 142, priority=3, callerid=0xb63e7b70 49711XXX, found=0xb5501208, combined_find_spawn=1) at pbx.c:3605 #20 0x081112ed in __ast_pbx_run (c=0xb63b7a90, args=0x0) at pbx.c:3692 #21 0x08112714 in pbx_thread (data=0xb63b7a90) at pbx.c:3965 #22 0x0816a6ed in dummy_start (data=0xb63e7dd0) at utils.c:861 #23 0xb7c9c4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #24 0xb7d9749e in clone () from /lib/tls/i686/cmov/libc.so.6 08. Dec #0 0xb8000424 in __kernel_vsyscall () #1 0xb7d2d6d0 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d2f098 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7d6b24d in ?? () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7d71604 in ?? () from /lib/tls/i686/cmov/libc.so.6 #5 0xb7d6d57d in _IO_file_seekoff () from /lib/tls/i686/cmov/libc.so.6 #6 0xb7d63760 in ?? () from /lib/tls/i686/cmov/libc.so.6 #7 0xb7d6ad37 in ftello64 () from /lib/tls/i686/cmov/libc.so.6 #8 0xb715cc19 in wav_read (s=0x8f08d78, whennext=0xb6adba14) at format_wav.c:352 #9 0x080dcebb in read_frame (s=0x8f08d78, whennext=0xb6adba14) at file.c:697 #10 0x080dcf48 in ast_readframe (s=0x8f08d78) at file.c:718 #11 0xb76508b3 in spawn_mp3 (class=0xb6ad6a34) at res_musiconhold.c:504 #12 0xb7650909 in spawn_mp3 (class=0x0) at res_musiconhold.c:511 #13 0x0809abc0 in ast_read_generator_actions (chan=0xb650a2d8, f=0xb650b810) at channel.c:2514 #14 0x0809c9e3 in __ast_read (chan=0xb650a2d8, dropaudio=0) at channel.c:3001 #15 0x0809cd50 in ast_read (chan=0xb650a2d8) at channel.c:3037 #16 0x08097743 in ast_safe_sleep_conditional (chan=0xb650a2d8, ms=1887, cond=0, data=0x0) at channel.c:1297 #17 0x080977a2 in ast_safe_sleep (chan=0xb650a2d8, ms=1) at channel.c:1309 #18 0xb765214c in moh_alloc (chan=0xb650a2d8, params=0xb6ade0b8) at res_musiconhold.c:905 #19 0x08107446 in pbx_exec (c=0xb650a2d8, app=0xb7a16d28, data=0xb6ade0b8) at pbx.c:951 #20 0x0810ee3f in pbx_extension_helper (c=0xb650a2d8, con=0x0, context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3, label=0x0, callerid=0xb6509a58 49711XXX, action=E_SPAWN, found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3138 #21 0x08110b76 in ast_spawn_extension (c=0xb650a2d8, context=0xb650a520 Click2Call4_0, exten=0xb650a570 142, priority=3, callerid=0xb6509a58 49711XXX, found=0xb6ae0208, combined_find_spawn=1) at pbx.c:3605 #22 0x081112ed in __ast_pbx_run (c=0xb650a2d8, args=0x0) at pbx.c:3692 #23 0x08112714 in pbx_thread (data=0xb650a2d8) at pbx.c:3965 #24 0x0816a6ed in dummy_start (data=0xb6502e98) at utils.c:861 #25 0xb7ceb4ff in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #26 0xb7de649e in
[asterisk-users] AudioCodes Caller ID
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have pstn-5665, digest has pstn-1270 [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user KMIEC Z sip:7804715...@10.0.0.157;tag=1c354211286 Calls go through but not Caller ID. Any suggestions? -- Joseph ___ -- 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] AudioCodes Caller ID
Joseph wrote: I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have pstn-5665, digest has pstn-1270 [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user KMIEC Z sip:7804715...@10.0.0.157;tag=1c354211286 Calls go through but not Caller ID. Any suggestions? Asterisk does not fully support domain authentication yet, so the 'username' present in the From header is used for authentication *and* Caller ID. That means that if you want proper Caller ID to be extracted from a SIP INVITE, you can't request authentication on that INVITE. If you configure the SIP user/peer that you are using for that gateway with 'insecure=invite', Asterisk will accept INVITEs from it without requiring authentication. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com 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
[asterisk-users] Daily Thousands of files in recording calls in Device mode
I installed an Elastix based system and changed it to work in Device-Mode since there is a call center and users has to login. As requested, I made recording always to all the users. The problem is there are no links in the Monitoring reports to the calls and while checking /var/spool/asterisk/monitor I found that there are thousands of 1 byte files for every day (!) and now we are already with 700,000 files after two weeks and actually without recordings. Also must mentions that there are OUT- files, but no IN.. Any ideas ? ___ -- 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] Daily Thousands of files in recording calls in Device mode
On Thu, Dec 31, 2009 at 12:12:19PM -0800, Yuval Yogev wrote: I installed an Elastix based system and changed it to work in Device-Mode That's FreePBX terminology. since there is a call center and users has to login. As requested, I made recording always to all the users. The problem is there are no links in the Monitoring reports to the calls and while checking /var/spool/asterisk/monitor I found that there are thousands of 1 byte files for every day (!) and now we are already with 700,000 files after two weeks and actually without recordings. I guess it;s time to look at the dialplan. Can you provide a trace of such a call? How many calls do you have each day? Also must mentions that there are OUT- files, but no IN.. Any ideas ? -- 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] AudioCodes Caller ID
On 12/31/09 13:06, Kevin P. Fleming wrote: Joseph wrote: I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have pstn-5665, digest has pstn-1270 [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user KMIEC Z sip:7804715...@10.0.0.157;tag=1c354211286 Calls go through but not Caller ID. Any suggestions? Asterisk does not fully support domain authentication yet, so the 'username' present in the From header is used for authentication *and* Caller ID. That means that if you want proper Caller ID to be extracted from a SIP INVITE, you can't request authentication on that INVITE. If you configure the SIP user/peer that you are using for that gateway with 'insecure=invite', Asterisk will accept INVITEs from it without requiring authentication. I've tried in sip.conf insecure=invite with user and peer still the same error and caller ID is not extracted. [pstn-1270] ; incoming/outgoing calls on FXO port 479-1270 type=peer secret=xxx username=pstn-5665 insecure=invite host=dynamic disallow=all allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup=1 Looking at this post it might be a bug in asterisk 1.4 (I'm using 1.4.22.1) https://issues.asterisk.org/view.php?id=9044 -- Joseph ___ -- 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] AudioCodes Caller ID
On 12/31/09 13:06, Kevin P. Fleming wrote: Joseph wrote: I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have pstn-5665, digest has pstn-1270 [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user KMIEC Z sip:7804715...@10.0.0.157;tag=1c354211286 Calls go through but not Caller ID. Any suggestions? Asterisk does not fully support domain authentication yet, so the 'username' present in the From header is used for authentication *and* Caller ID. That means that if you want proper Caller ID to be extracted from a SIP INVITE, you can't request authentication on that INVITE. If you configure the SIP user/peer that you are using for that gateway with 'insecure=invite', Asterisk will accept INVITEs from it without requiring authentication. type=peer insecure=invite I get the same and in addition call is not even forwarded to asterisk, it just keeps ringing. [Dec 31 14:42:34] WARNING[13715]: chan_sip.c:8553 check_auth: username mismatch, have pstn-5665, digest has pstn-1270 [Dec 31 14:42:34] NOTICE[13715]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user sip:pstn-1...@10.0.0.157;tag=1c1796801183 -- Joseph ___ -- 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] AudioCodes MWI
When I configured AudioCodes MP-114 to MWI it keeps complaining bout subscription without mailbox: chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer without mailbox: pstn-5665 chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer without mailbox: pstn-1270 these ports are FXO lines and sip.conf does not have any mailbox line for these lines. Anybody know how to turn off MWI for specific lines, is it done via MediaPack configuration or Asterisk sip.conf? -- Joseph ___ -- 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
