Re: [asterisk-users] Smallest possible Asterisk VM
On Tue, 2 Feb 2010, Frank Church wrote: How small can an Asterisk system be, in terms of disk space utilized? If depends on how much effort you're willing to put into making it small. This is one of my systems: FilesystemSize Used Avail Use% Mounted on /dev/ram0 136M 84M 53M 62% / So about 80MB, I know I can get it a lot smaller, but there's most of a text-based Linux install there too - a fairly full /bin, /usr/bin, etc. No *sql though - I've no need for it, but there is a full apache php. Media files are stored elsewhere and I'm a bit lazy sbout that: FilesystemSize Used Avail Use% Mounted on /dev/hda3 189M 95M 94M 51% /data that's the full set of sounds, MOH in all formats except WAV. If I could be bothered, I could work out exactly which ones are used and dump the rest, but I don't think there's a need. This system has a single 256MB compact flash device and 256MB of RAM, of which just over half is given up for the root filing system ramdisk. http://unicorn.drogon.net/cutie.jpg Gordon -- _ -- 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] Smallest possible Asterisk VM
On Tue, Feb 02, 2010 at 05:41:20AM +, Frank Church wrote: How small can an Asterisk system be, in terms of disk space utilized? Is disk-space your real issue? Disk space is cheap. Memory isn't that cheap, typically, on a VM environment. Likewise CPU usage. If you're on a hosted environment, you may also be limited on network bandwidth. I can give you some rough idea from the sie of our live CDs[1]. They use a compressed filesystem, so should generally assume that on a non-compressed file-system the disk usage is double[2]. A rather minimal install of Debian Lenny is some 110MB. This is a basic and non-optimized installation. Our live CD takes some 200MB and also includes Asterisk, PERL, Python (we needed it for AjaxTerm), Apache, and a bunch of other things. We didn't attempt to optimize it for space. We did not include a build envirnment, though. Other distributions tend to be much more desktop by default and thus need a specialized minimal version. This is often called JEOS (Just Enough OS). Such versions of at least Ubuntu and SuSE are available. My point is that maintainability is an important factor. What is your procedure for upgrading Asterisk? Upload a whole new image? This might be doable, e.g. if you set up a spare VM in advance. But you should be aware of this issue. I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? I suspect that you can easily get to 500MB with just about any distribution. Getting to 300MB and beyond may require some effort. Can packages be used to build the whole system, like using debs and rpms alone? [1] Our live CDs are based on Debian live. Debian-Live attempts to remain as close as possible to Debian proper. Their modifications to the original system, besides actually compressing the file system, are negligable (in terms of disk space usage). [2] You could use a compressed file system to save disk space. That would be trading disk space for CPU usage. It also means that disk access becomes slower, and I suspect this may hurt Asterisk. -- 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] uri tel: instead of sip:accepted ?
Hello all, Does asterisk accept uri tel: instead of sip: ? 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
Re: [asterisk-users] Smallest possible Asterisk VM
I have developed a minimal call shop billing system that includes an Asterisk VM and I want it to be as small as to reduce the installation size. 100Mb is good On 2 February 2010 05:41, Frank Church voi...@googlemail.com wrote: How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists -- _ -- 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 1.6.2 ?
Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? 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
Re: [asterisk-users] Asterisk 1.6.2 ?
On Tue, Feb 2, 2010 at 12:40 PM, hadi motamedi motamed...@gmail.com wrote: Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? Thank you Sorry . Forgot to mention that I have made use of the following packages for the upgrade procedure : asterisk-1.6.2.1.tar.gz dahdi-linux-complete-2.2.1+2.2.1.tar.gz libpri-1.4.10.2.tar.gz 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
[asterisk-users] Asterisk 1.6.1.13 and T.38 faxing
Hello everyone.
I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID
provider here in Brazil (GVT - Vox IP service). Here's my scenario:
When faxes arrive by a specific DID, they are routed thru this simple macro:
[macro-recebefax]
exten = s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1])
exten = s,n,Set(FAXCOUNT=${DB(fax/count)})
exten = s,n,Set(FAXFILE=fax-${DB(fax/count)}-rx)
exten = s,n,Answer()
exten = s,n,Wait(3)
exten = s,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE}.tif)
exten = s,n,NoOp(FAXSTATUS = ${FAXSTATUS})
exten = s,n,NoOp(FAXERROR = ${FAXERROR})
exten = s,n,NoOp(CALLID = ${CALLERID(name)} ${CALLERID(num)}
${REMOTESTATIONID})
exten = s,n,NoOp(FAXPAGES = ${FAXPAGES})
exten = s,n,NoOp(FAXBITRATE = ${FAXBITRATE})
exten = s,n,NoOp(FAXRESOLUTION = ${FAXRESOLUTION})
exten = s,n,NoOp(FAXMODE = ${FAXMODE})
exten = h,1,System(tiff2pdf -o /var/spool/asterisk/fax/${FAXFILE}.pdf -p A4
/var/spool/asterisk/fax/${FAXFILE}.tif)
exten = h,n,System(rm /var/spool/asterisk/fax/${FAXFILE}.tif)
exten = h,n,System(echo Fax recebido. /tmp/${FAXFILE}.txt)
exten = h,n,System(echo Remetente: ${CALLID} /tmp/${FAXFILE}.txt)
exten = h,n,System(echo Paginas: ${FAXPAGES} /tmp/${FAXFILE}.txt)
exten = h,n,System(echo Velocidade de transmissao: ${FAXBITRATE} bps
/tmp/${FAXFILE}.txt)
exten = h,n,System(echo Resolucao: ${FAXRESOLUTION} /tmp/${FAXFILE}.txt)
exten = h,n,System(mutt -s Allvo FAX -a
/var/spool/asterisk/fax/${FAXFILE}.pdf vinic...@canall.com.br
/tmp/${FAXFILE}.txt)
exten = h,n,System(rm /tmp/${FAXFILE}.txt)
I'm using here app_fax that comes with Asterisk, not the res_fax and
res_fax_digium that comes with FFA.
What happens is sometimes the T.38 negotiation goes well and others it fails
completely. That's what I got from the debug info on two different calls,
without changing any configs:
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP o=PVG 1265107050040 1265107050040 IN IP4 10.152.0.164...
UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP s=-... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP p=+1 613555... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP c=IN IP4 10.152.0.164... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP t=0 0... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (audio) SDP a=ptime:20... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8289 process_sdp_a_image:
FaxVersion: 0
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (image) SDP a=T38FaxVersion:0... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8263 process_sdp_a_image:
MaxBufferSize:1100
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (image) SDP a=T38FaxMaxBuffer:1100... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8298 process_sdp_a_image:
FaxMaxDatagram: 612
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (image) SDP a=T38FaxMaxDatagram:612... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8266 process_sdp_a_image:
T38MaxBitRate: 14400
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (image) SDP a=T38MaxBitRate:14400... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8335 process_sdp_a_image:
RateManagement: transferredTCF
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing
media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK.
[Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:8342 process_sdp_a_image: UDP EC:
t38UDPRedundancy
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164...
UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP s=-... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP p=+1 613555... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP c=IN IP4 10.152.0.164... OK.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp:
Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing
Vinícius Fontes wrote: [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP p=+1 613555... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP c=IN IP4 10.152.0.164... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=sqn: 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cdsc: 1 image udptl t38... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:4653 change_t38_state: T38 state changed to 0 on channel SIP/voxip- [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7932 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7937 process_sdp: We have an owner, now see if we need to change this call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:5231 update_call_counter: Updating call counter for incoming call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:3172 __sip_xmit: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.150.65.16:5060 [Feb 2 08:38:06] DEBUG[21064]: app_fax.c:699 transmit_t38: Shut down T.38 on SIP/voxip- Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on another one. Could that be a bug? I can show the entire SIP conversations if that's necessary for debugging this. That's not quite correct; in the second example, the T38 parameters are being sent as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not support. The second example does not provide backwards compatibility for SIP endpoints that do not support capability-based negotiation, whereas the first one does. -- 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] Asterisk 1.6.1.13 and T.38 faxing
Hi Kevin, On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: Vinícius Fontes wrote: [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP p=+1 613555... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP c=IN IP4 10.152.0.164... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=sqn: 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cdsc: 1 image udptl t38... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:4653 change_t38_state: T38 state changed to 0 on channel SIP/voxip- [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7932 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7937 process_sdp: We have an owner, now see if we need to change this call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:5231 update_call_counter: Updating call counter for incoming call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:3172 __sip_xmit: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.150.65.16:5060 [Feb 2 08:38:06] DEBUG[21064]: app_fax.c:699 transmit_t38: Shut down T.38 on SIP/voxip- Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on another one. Could that be a bug? I can show the entire SIP conversations if that's necessary for debugging this. That's not quite correct; in the second example, the T38 parameters are being sent as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not support. The second example does not provide backwards compatibility for SIP endpoints that do not support capability-based negotiation, whereas the first one does. What do you mean by that? Surely if you don't understand the cdsc and cpar lines you are supposed to simply ignore them, and carry on. If it were harmless, that capability information seems enormously useful, especially in the context of the current discussions on sorting out the mess that T.38 has become. Sadly, it does cause some systems to choke, which is probably why it is rarely included. Steve -- _ -- 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] Issue when reloading
Hello list! I´m having an issue when reloading Asterisk, I´ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a reload in Asterisk CLI and this is the output: isb152*CLI reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found == Parsing '/etc/asterisk/manager_additional.conf': == Found == Parsing '/etc/asterisk/manager_custom.conf': == Found == Parsing '/etc/asterisk/logger.conf': == Found Asterisk Event Logger restarted Asterisk Queue Logger restarted == Parsing '/etc/asterisk/features.conf': == Found == Parsing '/etc/asterisk/features_general_additional.conf': == Found == Parsing '/etc/asterisk/features_general_custom.conf': == Found == Parsing '/etc/asterisk/features_applicationmap_additional.conf': == Found == Parsing '/etc/asterisk/features_applicationmap_custom.conf': == Found == Parsing '/etc/asterisk/features_featuremap_additional.conf': == Found == Parsing '/etc/asterisk/features_featuremap_custom.conf': == Found -- Added extension '70' priority 1 to parkedcalls (0xa8798b0) -- Reloading module 'res_phoneprov' (HTTP Phone Provisioning) == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/sip_general_additional.conf': == Found == Parsing '/etc/asterisk/sip_general_custom.conf': == Found == Parsing '/etc/asterisk/sip_nat.conf': == Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found == Parsing '/etc/asterisk/sip_registrations.conf': == Found == Parsing '/etc/asterisk/sip_custom.conf': == Found == Parsing '/etc/asterisk/sip_additional.conf': == Found == Parsing '/etc/asterisk/sip_custom_post.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/phoneprov.conf': == Found -- Reloading module 'res_odbc' (ODBC resource) == Parsing '/etc/asterisk/res_odbc.conf': == Found -- Reloading module 'res_musiconhold' (Music On Hold Resource) -- Reloading module 'res_crypto' (Cryptographic Digital Signatures) -- Reloading module 'res_config_odbc' (Realtime ODBC configuration) -- Reloading module 'res_clialiases' (CLI Aliases) -- Reloading module 'res_adsi' (ADSI Resource) -- Reloading module 'pbx_dundi' (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': == Found -- Reloading module 'pbx_config' (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': == Found == Parsing '/etc/asterisk/extensions_override_freepbx.conf': == Found == Parsing '/etc/asterisk/extensions_additional.conf': == Found == Parsing '/etc/asterisk/globals_custom.conf': == Found == Parsing '/etc/asterisk/extensions_custom.conf': == Found [Feb 2 08:14:46] NOTICE[32490]: pbx_ael.c:149 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. -- Reloading module 'func_odbc' (ODBC lookups) -- Reloading module 'codec_ulaw' (mu-Law Coder/Decoder) -- Reloading module 'codec_lpc10' (LPC10 2.4kbps Coder/Decoder) -- Reloading module 'codec_gsm' (GSM Coder/Decoder) -- Reloading module 'codec_g726' (ITU G.726-32kbps G726 Transcoder) -- Reloading module 'codec_g722' (ITU G.722-64kbps G722 Transcoder) -- Reloading module 'codec_dahdi' (Generic DAHDI Transcoder Codec Translator) -- Reloading module 'codec_alaw' (A-law Coder/Decoder) -- Reloading module 'codec_adpcm' (Adaptive Differential PCM Coder/Decoder) -- Reloading module 'chan_unistim' (UNISTIM Protocol (USTM)) Reloading unistim.conf... == Parsing '/etc/asterisk/unistim.conf': == Found -- Reloading module 'chan_skinny' (Skinny Client Control Protocol (Skinny)) [Feb 2 08:14:46] NOTICE[32490]: chan_skinny.c:7062 config_load: Configuring skinny from skinny.conf == Parsing '/etc/asterisk/skinny.conf': == Found -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Nothing about chan_dahdi, app_queue, cdr or chan_agent But, for example if I change something in agents.conf (anything), the output of the reload in CLI shows me that chan_dahdi, app_queue, chan_agents, etc now reloads its configuration. isb152*CLI reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found == Parsing '/etc/asterisk/manager_additional.conf': == Found == Parsing '/etc/asterisk/manager_custom.conf': == Found == Parsing '/etc/asterisk/logger.conf': == Found Asterisk Event Logger restarted Asterisk Queue Logger restarted == Parsing '/etc/asterisk/features.conf': == Found == Parsing '/etc/asterisk/features_general_additional.conf': == Found == Parsing '/etc/asterisk/features_general_custom.conf': == Found == Parsing
Re: [asterisk-users] Problems with recordings of call using Monitor
thank you for that tip:
*exten =
_XXX,1,Set(MONITOR_FILENAME=naar-${EXTEN}-van-${CALLERID(number)}-wanneer-${STRFTIME(${EPOCH},,%Y%m%d-%H:%M:%S)}-uniekid-${UNIQUEID}.wav,b)
*
*exten = _XXX,n,MixMonitor(${MONITOR_FILENAME})*
works great!
On Mon, Feb 1, 2010 at 8:39 PM, David Backeberg dbackeb...@gmail.comwrote:
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
I'm using the default Asterisk function Monitor to record calls, but i
have
some issue's with this, the problem is when a call is finished, it never
mix
in out together, bellow you can see my call configuration:
Perhaps you would prefer to use MixMonitor() rather than Monitor()
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Groet // Kind regards,
Peter den Hartog
Sent from Amsterdam, NH, Netherlands
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Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing
Steve Underwood wrote: Hi Kevin, On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: Vinícius Fontes wrote: [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP p=+1 613555... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP c=IN IP4 10.152.0.164... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=sqn: 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cdsc: 1 image udptl t38... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:4653 change_t38_state: T38 state changed to 0 on channel SIP/voxip- [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7932 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7937 process_sdp: We have an owner, now see if we need to change this call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:5231 update_call_counter: Updating call counter for incoming call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:3172 __sip_xmit: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.150.65.16:5060 [Feb 2 08:38:06] DEBUG[21064]: app_fax.c:699 transmit_t38: Shut down T.38 on SIP/voxip- Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on another one. Could that be a bug? I can show the entire SIP conversations if that's necessary for debugging this. That's not quite correct; in the second example, the T38 parameters are being sent as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not support. The second example does not provide backwards compatibility for SIP endpoints that do not support capability-based negotiation, whereas the first one does. What do you mean by that? Surely if you don't understand the cdsc and cpar lines you are supposed to simply ignore them, and carry on. If it were harmless, that capability information seems enormously useful, especially in the context of the current discussions on sorting out the mess that T.38 has become. Sadly, it does cause some systems to choke, which is probably why it is rarely included. In this case, the re-INVITE did *not* include a non-capabilities-based offer for T.38. The SDP parser listed the cdsc and cpar lines as unsupported, but it did not see any media stream offer for T.38 (unlike the OP's first example), so it set the internal T.38 state on the channel to 'not in use'. -- 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] Asterisk 1.6.1.13 and T.38 faxing
On 02/02/2010 10:11 PM, Kevin P. Fleming wrote: Steve Underwood wrote: Hi Kevin, On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: Vinícius Fontes wrote: [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP p=+1 613555... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP c=IN IP4 10.152.0.164... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=sqn: 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cdsc: 1 image udptl t38... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:4653 change_t38_state: T38 state changed to 0 on channel SIP/voxip- [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7932 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7937 process_sdp: We have an owner, now see if we need to change this call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:5231 update_call_counter: Updating call counter for incoming call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:3172 __sip_xmit: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.150.65.16:5060 [Feb 2 08:38:06] DEBUG[21064]: app_fax.c:699 transmit_t38: Shut down T.38 on SIP/voxip- Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on another one. Could that be a bug? I can show the entire SIP conversations if that's necessary for debugging this. That's not quite correct; in the second example, the T38 parameters are being sent as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not support. The second example does not provide backwards compatibility for SIP endpoints that do not support capability-based negotiation, whereas the first one does. What do you mean by that? Surely if you don't understand the cdsc and cpar lines you are supposed to simply ignore them, and carry on. If it were harmless, that capability information seems enormously useful, especially in the context of the current discussions on sorting out the mess that T.38 has become. Sadly, it does cause some systems to choke, which is probably why it is rarely included. In this case, the re-INVITE did *not* include a non-capabilities-based offer for T.38. The SDP parser listed the cdsc and cpar lines as unsupported, but it did not see any media stream offer for T.38 (unlike the OP's first example), so it set the internal T.38 state on the channel to 'not in use'. That's how T.38 calls normally start. They mostly start as audio, and switch into T.38 mode later. We have only seen an initial fragment in the log. We haven't seen anything that's actually wrong. We see an offer to do telephony events, and from there things might progress to T.38 or something else. I can't see anything invalid in that, even if the cdsc/cpar stuff is not understood. Steve -- _ -- 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] NVFaxDetect
I've followed these procedures before and they've worked fine. http://www.freepbx.org/forum/freepbx/tips-and-tricks/freepbx-2-6-nvfax-detection-asterisk-1-6-1-6-and-digium-fax-working On Mon, Feb 1, 2010 at 2:15 PM, Danny Nicholas da...@debsinc.com wrote: YMMV, but NVFaxdetect can be activated in 1.4.x; Try this link http://blogtech.oc9.com/index.php?option=com_contentview=articleid=77%3A20 071121astItemid=8http://blogtech.oc9.com/index.php?option=com_contentview=articleid=77%3A20%0A071121astItemid=8 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Monday, February 01, 2010 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NVFaxDetect On 02/01/10 12:01, Kevin P. Fleming wrote: Joseph wrote: I was running it with Asterisk-1.2 (nice application) one would think that it should be incorporate into asterisk but they are not doing it, nor the developer is interesting in upgrading the code ever-time there is a new version of asterisk and something is broken. I was not able to install it with version 1.4 It has never been offered for inclusion into Asterisk, so they cannot do it. The Asterisk project does not 'pull in' code, it must be specifically offered for inclusion by the author(s)/copyright holder(s) of the code. -- 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 Thank for explanation, it clears things up. It is/was a nice application and I'm just wandering why develp a code and let it it die -- 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 -- _ -- 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 1.6.1.13 and T.38 faxing
Steve Underwood wrote: That's how T.38 calls normally start. They mostly start as audio, and switch into T.38 mode later. We have only seen an initial fragment in the log. We haven't seen anything that's actually wrong. We see an offer to do telephony events, and from there things might progress to T.38 or something else. I can't see anything invalid in that, even if the cdsc/cpar stuff is not understood. Umm, yeah. I knew all that :-) Since the OP is using ReceiveFAX in his dialplan, that means Asterisk sent the T.38 re-INVITE. Under the assumption that's he posted the snippets from corresponding sections of successful and failed calls, the SDP parsing output we saw would be from the response to the re-INVITE. In the second example, the response to the re-INVITE did not include a T.38 media stream. I agree, though, that a more complete log would be most helpful in clarifying the situation. -- 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] Use of 603 Declined
On Sat, Jan 30, 2010 at 9:03 AM, Kevin P. Fleming kpflem...@digium.com wrote: Olle E. Johansson wrote: Here's something interesting: 21.4.25 487 Request Terminated The request was terminated by a BYE or CANCEL request. This response is never returned for a CANCEL request itself. This is only used in combination with Cancel's, but in this case the call was cancelled by the dialplan, not by the caller. It's a misuse, but a bit clever one. Yes, I think 487 seems to be a logical choice here; it's very close to what 487 is normatively used for. Now, I realize that we also need a setting to indicate whether Asterisk is authorative for a domain or not. If we're the only owners of a domain, we should generate 6xx class errors, if not, 4xx error. So this also applies to 486/600 busy 488/606 and 404/604. If we start separating 4xx and 6xx replies, we might as well do it right. So the domain configuration needs an option per domain whether we're just part of a cluster handling a domain or if we're THE domain handler. That would be a good idea, yes. The quick fix for my issue is to make Asterisk send 487. The real fix, however, looks to be much broader (configuration option for domains, etc). How much work is this? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] Asterisk 1.6.1.13 and T.38 faxing
- Steve Underwood ste...@coppice.org escreveu: On 02/02/2010 10:11 PM, Kevin P. Fleming wrote: Steve Underwood wrote: Hi Kevin, On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: Vinícius Fontes wrote: [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP s=-... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP p=+1 613555... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP c=IN IP4 10.152.0.164... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=sqn: 0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cdsc: 1 image udptl t38... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxVersion:0... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP a=cpar: a=T38FaxUdpEC:t38UDPRedundancy... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7753 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK. [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:4653 change_t38_state: T38 state changed to 0 on channel SIP/voxip- [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7932 process_sdp: We're settling with these formats: 0x8 (alaw) [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7937 process_sdp: We have an owner, now see if we need to change this call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:5231 update_call_counter: Updating call counter for incoming call [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:3172 __sip_xmit: Trying to put 'ACK sip:10.' onto UDP socket destined for 10.150.65.16:5060 [Feb 2 08:38:06] DEBUG[21064]: app_fax.c:699 transmit_t38: Shut down T.38 on SIP/voxip- Note how items like T38FaxUdpEC are listed as OK on one call and unsupported on another one. Could that be a bug? I can show the entire SIP conversations if that's necessary for debugging this. That's not quite correct; in the second example, the T38 parameters are being sent as 'capabilities' (a=cdsc and a=cpar) which Asterisk does not support. The second example does not provide backwards compatibility for SIP endpoints that do not support capability-based negotiation, whereas the first one does. What do you mean by that? Surely if you don't understand the cdsc and cpar lines you are supposed to simply ignore them, and carry on. If it were harmless, that capability information seems enormously useful, especially in the context of the current discussions on sorting out the mess that T.38 has become. Sadly, it does cause some systems to choke, which is probably why it is rarely included. In this case, the re-INVITE did *not* include a non-capabilities-based offer for T.38. The SDP parser listed the cdsc and cpar lines as unsupported, but it did not see any media stream offer for T.38 (unlike the OP's first example), so it set the internal T.38 state on the channel to 'not in use'. That's how T.38 calls normally start. They mostly start as audio, and switch into T.38 mode later. We have only seen an initial fragment in the log. We haven't seen anything that's actually wrong. We see an offer to do telephony events, and from there things might progress to T.38 or something else. I can't see anything invalid in that, even if the cdsc/cpar stuff is not understood. Steve I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. -- _ -- 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 1.6.1.13 and T.38 faxing
Vinícius Fontes wrote: I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. Post a 'sip set debug' capture of the failing call in this thread; that will make it much more obvious what is happening. -- 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] Astribank problem
I apologize, but what benefit does Astribank provide over something like a Quintum series gateway with FXS/FXO support? Whats the benefit of USB vs Ethernet? --Dovey -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *frangky robert *Sent:* Friday, January 29, 2010 10:58 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Astribank problem H all... I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final. My problem is, every time i unplug the astribank power supply, and reconnect it, astribank cannot work again (lsusb result is 11x0)... but, after reinstall the asterisk and dahdi, astribank will detected (lsusb result is 11x2)... any suggestion? Regard, frank. -- Chat online and in real-time with friends and family! Windows Live Messengerhttp://get.live.com/messenger/overview -- _ -- 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] Astribank problem
On Tue, Feb 02, 2010 at 11:16:13AM -0500, Dovey Forman wrote: I apologize, but what benefit does Astribank provide over something like a Quintum series gateway with FXS/FXO support? Whats the benefit of USB vs Ethernet? One simple benefit: it's on the same system, and does not require separate configuration. If you don't want it, I suppose you won't buy it :-) -- 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] Asterisk 1.6.1.13 and T.38 faxing
- Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. Post a 'sip set debug' capture of the failing call in this thread; that will make it much more obvious what is happening. -- 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 I've put it on pastebin because is was a lot of text. Here's the link: http://pastebin.com/m7467cea1. That's all the information on the CLI with verbose=3 and sip set debug peer voxip. -- _ -- 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] Astribank problem
You'll find out the benefit when you change anything on your dialplan, making it necessary to alter the dialplan on every FXS port of your gateway. :) Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 - Tzafrir Cohen tzafrir.co...@xorcom.com escreveu: On Tue, Feb 02, 2010 at 11:16:13AM -0500, Dovey Forman wrote: I apologize, but what benefit does Astribank provide over something like a Quintum series gateway with FXS/FXO support? Whats the benefit of USB vs Ethernet? One simple benefit: it's on the same system, and does not require separate configuration. If you don't want it, I suppose you won't buy it :-) -- 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 -- _ -- 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] Two Extensions showing as Busy
Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running Asterisk 1.6.1.12 and when connecting to the console we saw :- [Feb 1 09:18:38] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577921) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:40] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577923) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:42] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577925) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite XXX*CLI sip show channels Peer User/ANRCall ID Format Hold Last MessageExpiry 10.XX.X.XXX USER1 214d5e15267a17f 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX (none) 2071025406 0x0 (nothing)No Rx: REGISTER 10.XX.X.XXX USER2 3c711a8f0456-oa 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER3 0bbd02e95245bda 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER4 3c64d1451cbb-dr 0x4 (ulaw) No Rx: ACK 5 active SIP dialogs [Feb 1 09:18:43] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577927) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:45] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577929) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:47] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577931) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:49] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577933) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite All extensions were within the same context. Has anybody seen this behavior before and what could cause it ? we resolved the issue by sending a SIP notify reboot to the SNOM phones. Thanks, Phil -- _ -- 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] Astribank problem
What do you mean same system? Isnt it separate hardware connected via USB? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, February 02, 2010 11:38 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Astribank problem On Tue, Feb 02, 2010 at 11:16:13AM -0500, Dovey Forman wrote: I apologize, but what benefit does Astribank provide over something like a Quintum series gateway with FXS/FXO support? Whats the benefit of USB vs Ethernet? One simple benefit: it's on the same system, and does not require separate configuration. If you don't want it, I suppose you won't buy it :-) -- 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 -- _ -- 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] Codec coversion
Hi: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated. Thanks -- _ -- 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] Two Extensions showing as Busy
Could be a call-limit setting problem (just a WAG) -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Tuesday, February 02, 2010 11:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Two Extensions showing as Busy Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running Asterisk 1.6.1.12 and when connecting to the console we saw :- [Feb 1 09:18:38] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577921) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:40] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577923) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:42] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577925) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite XXX*CLI sip show channels Peer User/ANRCall ID Format Hold Last MessageExpiry 10.XX.X.XXX USER1 214d5e15267a17f 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX (none) 2071025406 0x0 (nothing)No Rx: REGISTER 10.XX.X.XXX USER2 3c711a8f0456-oa 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER3 0bbd02e95245bda 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER4 3c64d1451cbb-dr 0x4 (ulaw) No Rx: ACK 5 active SIP dialogs [Feb 1 09:18:43] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577927) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:45] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577929) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:47] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577931) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:49] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577933) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite All extensions were within the same context. Has anybody seen this behavior before and what could cause it ? we resolved the issue by sending a SIP notify reboot to the SNOM phones. Thanks, Phil -- _ -- 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] Two Extensions showing as Busy
- Danny Nicholas da...@debsinc.com wrote: Could be a call-limit setting problem (just a WAG) -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Tuesday, February 02, 2010 11:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Two Extensions showing as Busy Hi, had a very strange issue today where two extensions called each other, hung up, but on another extension the two parties continued to show as busy. This was running Asterisk 1.6.1.12 and when connecting to the console we saw :- [Feb 1 09:18:38] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577921) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:40] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577923) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:42] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577925) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite XXX*CLI sip show channels Peer User/ANRCall ID Format Hold Last MessageExpiry 10.XX.X.XXX USER1 214d5e15267a17f 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX (none) 2071025406 0x0 (nothing)No Rx: REGISTER 10.XX.X.XXX USER2 3c711a8f0456-oa 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER3 0bbd02e95245bda 0x4 (ulaw) No Tx: ACK 10.XX.X.XXX USER4 3c64d1451cbb-dr 0x4 (ulaw) No Rx: ACK 5 active SIP dialogs [Feb 1 09:18:43] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577927) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:45] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577929) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:47] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577931) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite [Feb 1 09:18:49] WARNING[17474]: chan_sip.c:17046 handle_response_invite: just did sched_add waitid(577933) for sip_reinvite_retry for dialog 3c711a8f0456-oa3xwb6nxknk in handle_response_invite All extensions were within the same context. Has anybody seen this behavior before and what could cause it ? we resolved the issue by sending a SIP notify reboot to the SNOM phones. call-limit is deprecated in this version so I would have thought it would not effect it. -- Thanks, Phil -- _ -- 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 1.6.1.13 and T.38 faxing
- Vinícius Fontes vinic...@canall.com.br escreveu: - Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. Post a 'sip set debug' capture of the failing call in this thread; that will make it much more obvious what is happening. -- 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 UPDATE: I tested using the very same SIP configs with both app_fax and res_fax/res_fax_digium. The Fax For Asterisk solution works perfectly. app_fax on the other hand fails miserably every single time. I'm pretty sure it's an app_fax issue. Should I go ahead and post an issue on Mantis? -- _ -- 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 1.6.1.13 and T.38 faxing
On 02/03/2010 12:45 AM, Vinícius Fontes wrote: - Kevin P. Flemingkpflem...@digium.com escreveu: Vinícius Fontes wrote: I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. Post a 'sip set debug' capture of the failing call in this thread; that will make it much more obvious what is happening. -- 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 I've put it on pastebin because is was a lot of text. Here's the link: http://pastebin.com/m7467cea1. That's all the information on the CLI with verbose=3 and sip set debug peer voxip. I wonder why Asterisk would say: X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 344 v=0 o=root 44350963 44350964 IN IP4 10.153.66.146 s=Asterisk PBX 1.6.1.13 c=IN IP4 10.153.66.146 t=0 0 m=image 4819 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy I'm pretty sure it doesn't support T38FaxTranscodingMMR or T38FaxTranscodingJBIG, so they should not be there. Perhaps more relevant to you, though, is why is * saying (External RTP bridge). Does it really mean it? Steve -- _ -- 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] Help for MOH - sounding scratchy/static on hold
Thanks for all your inputs... Jeff: I used the converter to convert the sound file to 8KHz, 16 bit PCM encoded mono wave file which asterisk needs and now the POTS call quality is lot clear than before but the cell phone is still the same, not much clear...i think because of its voice codec as you mentioned. Regards Sandesh On Sat, Jan 30, 2010 at 10:38 AM, hin lee hi...@yahoo.com wrote: I am also having this issue with the MOH. Would be nice to find a solution! -- *From:* Steve Edwards asterisk@sedwards.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Fri, January 29, 2010 3:43:12 PM *Subject:* Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to process more calls. Are you in a severely storage challenged environment? You should store all of your audio encoded to match the codec used by the channel. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Codec coversion
On Tue, 2 Feb 2010, wassim darwich wrote: Is there any software or hadware for codec conversion on asterisk ,any suggestion will be appreciated. Your question is not specific enough for a specific answer. If you are looking for a command line utility that will transcode a file from one codec (wav) to another codec (ulaw), sox would be my preference. Newer versions of Asterisk (I still use 1.2) have an Asterisk command line command that will transcode a file. This strikes me like using a sledge hammer to drive a tack. Also, I'd hate to have to explain to my boss that a bug in my transcode a file script took out the PBX. If you are asking if Asterisk can transcode between 2 legs of a call -- yes. That's what all those codec_*.so files are for. If you have a codec file for each leg, Asterisk will do the magic in between. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help for MOH - sounding scratchy/static on hold
Sandesh- I used the converter to convert the sound file to 8KHz, 16 bit PCM encoded mono wave file which asterisk needs and now the POTS call quality is lot clear than before but the cell phone is still the same, not much clear...i think because of its voice codec as you mentioned. Ok sounds good, hehe. -Jeff On Sat, Jan 30, 2010 at 10:38 AM, hin lee hi...@yahoo.com wrote: I am also having this issue with the MOH. Would be nice to find a solution! -- *From:* Steve Edwards asterisk@sedwards.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Fri, January 29, 2010 3:43:12 PM *Subject:* Re: [asterisk-users] Help for MOH - sounding scratchy/static on hold On Fri, 29 Jan 2010, Danny Nicholas wrote: Mpg123 works well for us. You have to get your files into mp3 format, but LAME does this simply. Why would you want to compress files when you will have to decompress them again every single time the are used? I'd rather use the CPU cycles to process more calls. Are you in a severely storage challenged environment? You should store all of your audio encoded to match the codec used by the channel. -- _ -- 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 1.6.1.13 and T.38 faxing
Steve Underwood wrote: I wonder why Asterisk would say: X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 344 v=0 o=root 44350963 44350964 IN IP4 10.153.66.146 s=Asterisk PBX 1.6.1.13 c=IN IP4 10.153.66.146 t=0 0 m=image 4819 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy I'm pretty sure it doesn't support T38FaxTranscodingMMR or T38FaxTranscodingJBIG, so they should not be there. Perhaps more relevant to you, though, is why is * saying (External RTP bridge). Does it really mean it? That latter part is just a small bug in chan_sip; any re-INVITE sent on a call gets that tag, because originally direct media path (external bridging) was the only means to generate re-INVITE requests. Now that T.38 can do it as well, the code hasn't been changed to properly tag them. As far as the T.38 parameters go, those are under control of the application that caused the re-INVITE (or the bridged channel, if this is a passthrough situation). If he's using app_fax/spandsp, I believe we currently have app_fax configured to offer TranscodingMMR and TranscodingJBIG because spandsp supports those modes. The Fax for Asterisk product does not, so re-INVITEs generated by that implementation would not include those options. -- 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] sip realtime md5secret
Hi all, Does asterisk cache realtime sip md5secret values? I create a user over a web site and set a password as asd and I can login with that password. After a while I change my password and set it as 123. Although the password is set as 123 in the mysql database (I double checked), i can not login using the password 123, but with asd. So, am i missing a point? or is this how asterisk works? and Should I reload asterisk after adding a peer in the database? Any help would be appreciated. -- Emre Kurnaz ITU/BIDB Sistem Destek Grubu RHCE : 805009174841679 Yarı Zamanlı Öğrenci Koordinatörü kurn...@itu.edu.tr 0212 285 3930 ITU Linux Academy http://ila.itu.edu.tr -- _ -- 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 1.6.1.13 and T.38 faxing
- Steve Underwood ste...@coppice.org escreveu: On 02/03/2010 12:45 AM, Vinícius Fontes wrote: - Kevin P. Flemingkpflem...@digium.com escreveu: Vinícius Fontes wrote: I couldn't agree more Steve. Is there any other info I could provide in order to help you find out what's wrong? I could even open an issue on Mantis if the Digium staff think it's worth it. Post a 'sip set debug' capture of the failing call in this thread; that will make it much more obvious what is happening. -- 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 I've put it on pastebin because is was a lot of text. Here's the link: http://pastebin.com/m7467cea1. That's all the information on the CLI with verbose=3 and sip set debug peer voxip. I wonder why Asterisk would say: X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 344 v=0 o=root 44350963 44350964 IN IP4 10.153.66.146 s=Asterisk PBX 1.6.1.13 c=IN IP4 10.153.66.146 t=0 0 m=image 4819 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy I'm pretty sure it doesn't support T38FaxTranscodingMMR or T38FaxTranscodingJBIG, so they should not be there. Perhaps more relevant to you, though, is why is * saying (External RTP bridge). Does it really mean it? Steve I'm not really sure. What I know is that this telco has separate boxes for SIP signalling and RTP media. Not even sure if that's related to your question which, to be honest, I didn't fully understand. :) -- _ -- 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 1.6.1.13 and T.38 faxing
Vinícius Fontes wrote: I've put it on pastebin because is was a lot of text. Here's the link: http://pastebin.com/m7467cea1. That's all the information on the CLI with verbose=3 and sip set debug peer voxip. OK, with the complete capture we can see that the problem is actually quite different. In this call, Asterisk sent a re-INVITE to T.38 mode from audio mode, the provider accepted it, and then Asterisk acknowledged it. Immediately afterwards, Asterisk sent a re-INVITE *back* to audio mode, which the provider accepted (and included T.38 capabilities in their response). Because of this, the FAX reception process failed since the T.38 session was destroyed. The most likely cause of this problem is a bug in chan_sip, but it has been fixed for quite some time now, and the fix is included in 1.6.1.13. This would also fit with your statement about not having this issue with Fax For Asterisk, as it does not generate any audio frames while negotiating T.38 as the receiver of a FAX. I would suggest opening an issue in the issue tracker at issues.asterisk.org and uploading your console trace there; there is clearly a bug here that needs to be found and fixed. -- 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] Asterisk 1.6.1.13 and T.38 faxing
On 02/03/2010 03:14 AM, Kevin P. Fleming wrote: Steve Underwood wrote: I wonder why Asterisk would say: X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 344 v=0 o=root 44350963 44350964 IN IP4 10.153.66.146 s=Asterisk PBX 1.6.1.13 c=IN IP4 10.153.66.146 t=0 0 m=image 4819 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy I'm pretty sure it doesn't support T38FaxTranscodingMMR or T38FaxTranscodingJBIG, so they should not be there. Perhaps more relevant to you, though, is why is * saying (External RTP bridge). Does it really mean it? That latter part is just a small bug in chan_sip; any re-INVITE sent on a call gets that tag, because originally direct media path (external bridging) was the only means to generate re-INVITE requests. Now that T.38 can do it as well, the code hasn't been changed to properly tag them. As far as the T.38 parameters go, those are under control of the application that caused the re-INVITE (or the bridged channel, if this is a passthrough situation). If he's using app_fax/spandsp, I believe we currently have app_fax configured to offer TranscodingMMR and TranscodingJBIG because spandsp supports those modes. The Fax for Asterisk product does not, so re-INVITEs generated by that implementation would not include those options. Spandsp doesn't support those features. I don't know anything which does. It seems they can only be used with TCP. Spandsp does support T38FaxFillBitRemoval which the FAX for Asterisk package does not (according to Commetrex). Steve -- _ -- 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] codec conversion
Hi: Thanks for your reply,ill give you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider only accept g723 ,So what i have to do is to receive g711 codec and convert them to g723 at asterisk ,i tried this before but i saw the cpu usage is overloaded when doing conversion ,Iam using asterisk 1.4.22 ,So what do you advice me. -- _ -- 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] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb
) in new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the
call to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments.
Side Note: I am James, Jim is my future father in law!
image001.jpg--
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Re: [asterisk-users] Semi-Transfer
What lines are in your group 3? It is possible that DAHDI/52 is the only
line in that group and that's why you're getting the all congested.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing that.
Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S
)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb) in
new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the call
to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.
cid:image003.png@01C9F268.65A4F5C0
Side Note: I am James, Jim is my future father in law!
image001.jpg--
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Re: [asterisk-users] Semi-Transfer
You might also consider the DISA command instead of Dial.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing that.
Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S
)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb) in
new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the call
to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.
cid:image003.png@01C9F268.65A4F5C0
Side Note: I am James, Jim is my future father in law!
image001.jpg--
_
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Re: [asterisk-users] Semi-Transfer
That is the PRI span there are many available lines.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer
What lines are in your group 3? It is possible that DAHDI/52 is the
only line in that group and that's why you're getting the all
congested.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb
) in new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the
call to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments.
Side Note: I am James, Jim is my future father in law!
image001.jpg--
_
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Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing
- Kevin P. Fleming kpflem...@digium.com escreveu: Vinícius Fontes wrote: I've put it on pastebin because is was a lot of text. Here's the link: http://pastebin.com/m7467cea1. That's all the information on the CLI with verbose=3 and sip set debug peer voxip. OK, with the complete capture we can see that the problem is actually quite different. In this call, Asterisk sent a re-INVITE to T.38 mode from audio mode, the provider accepted it, and then Asterisk acknowledged it. Immediately afterwards, Asterisk sent a re-INVITE *back* to audio mode, which the provider accepted (and included T.38 capabilities in their response). Because of this, the FAX reception process failed since the T.38 session was destroyed. The most likely cause of this problem is a bug in chan_sip, but it has been fixed for quite some time now, and the fix is included in 1.6.1.13. This would also fit with your statement about not having this issue with Fax For Asterisk, as it does not generate any audio frames while negotiating T.38 as the receiver of a FAX. I would suggest opening an issue in the issue tracker at issues.asterisk.org and uploading your console trace there; there is clearly a bug here that needs to be found and fixed. -- 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 Reported as issue 16576. Thanks for the support! Another related issue, and this one happens in FFA as well. I noticed I can only get really reliable fax reception if I edit chan_sip.c to force the bitrate down to 4800. Otherwise most of the times I get a few lines OK and the rest all garbage. Already talked to the telco tech support, they say there's no packet loss on their side (UDPTL isn't transmitted via Internet, my Asterisk box is connected to them using a dedicated VPN circuit), and I confirmed that with Wireshark. According to them, signalling is okay too. One thing I noticed is that Asterisk 1.6.1.13 completely ignores the maxdatagram setting on sip.conf. No matter what I set there, I keep getting the default value of 612 as offered by the telco. Asterisk not even once tries to negotiate that. Maybe (and that's a longshot) the datagram size is too long, or the buffers too low and Asterisk can't keep up with the reception of UDPTL packets? Is there any way to rule that out? As usual, I'm willing to provide any info to help solve this. -- _ -- 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] Semi-Transfer
This wiki is outdated but the group stuff still applies to DAHDI
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
Assuming that you have many available lines in group 3, changing the option
to g3 from G3 might help.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Semi-Transfer
That is the PRI span there are many available lines.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer
What lines are in your group 3? It is possible that DAHDI/52 is the only
line in that group and that's why you're getting the all congested.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing that.
Below is what I'm using in my extensions.conf
exten= X,1,Read(num,/var/lib/asterisk/sounds/mtas/10digit,10,,,5)
exten= X,2,SayDigits(${num})
exten= X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten= X,4,WaitExten(3)
exten=
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S
)}-${CALLERID(num)}-${EXTEN},mb)
exten= X,6,dial(${belltd}/${num})
Here is what I see in the CMD when the dial fails
-- Timeout on DAHDI/52-1, continuing...
-- Executing [xxx...@recout:5] Monitor(DAHDI/52-1,
wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb) in
new stack
-- Executing [XX @RecOut:6] Dial(DAHDI/52-1,
DAHDI/G3/4099819750) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G3/4099819750
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'
-- Hungup 'DAHDI/52-1'
Now all of my lines are NOT ties up so there is available paths for the call
to go out.
Anyway so how would I accomplish this transfer of sorts?
James Shigley
Monroe Telephone Answering Service
409-981-9750
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.
cid:image003.png@01C9F268.65A4F5C0
Side Note: I am James, Jim is my future father in law!
image001.jpg--
_
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Re: [asterisk-users] sip realtime md5secret
On Tue, 2010-02-02 at 21:20 +0200, Emre Kurnaz wrote: Hi all, Does asterisk cache realtime sip md5secret values? I create a user over a web site and set a password as asd and I can login with that password. After a while I change my password and set it as 123. Although the password is set as 123 in the mysql database (I double checked), i can not login using the password 123, but with asd. So, am i missing a point? or is this how asterisk works? and Should I reload asterisk after adding a peer in the database? Any help would be appreciated. If you have rtcachefriends=yes set in your sip.conf file then you either have to wait until the peer expires or you have to reload sip so the peer is re read from the database. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] codec conversion
On Tue, 2 Feb 2010, wassim darwich wrote: Thanks for?your reply,ill give?you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider?only accept g723 ,So what i have to do is to receive?g711?codec and convert them to g723 at?asterisk ,i tried this before but i saw the cpu?usage is overloaded when doing conversion ,Iam using asterisk 1.4.22 ,So what do you advice me. Get your client to switch to g723 or your provider to switch to ulaw. If that is not possible, get more CPU resources: 1) Eliminate any unrelated processes (X, Apache, MySQL, Tetris). Make sure Asterisk is running with elevated priority. 2) If your other processes (AGIs?) are written in scripting languages (Perl, PHP), re-code them in compiled languages (C). 3) Use more powerful processors (faster clock, more cores, more processors). 4) Split the load across multiple hosts. This has the added advantage of not putting all your eggs in one basket -- you can take a host out of service for maintenance or upgrades. 5) If you are swapping, more RAM may help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] codec conversion
On Tue, 2 Feb 2010, Steve Edwards wrote: On Tue, 2 Feb 2010, wassim darwich wrote: Thanks for?your reply,ill give?you my situation, iam using my asterisk box as a switch ,so my client is sending me ulaw and my voip provider?only accept g723 ,So what i have to do is to receive?g711?codec and convert them to g723 at?asterisk ,i tried this before but i saw the cpu?usage is overloaded when doing conversion ,Iam using asterisk 1.4.22 ,So what do you advice me. Get your client to switch to g723 or your provider to switch to ulaw. If that is not possible, get more CPU resources: 1) Eliminate any unrelated processes (X, Apache, MySQL, Tetris). Make sure Asterisk is running with elevated priority. 2) If your other processes (AGIs?) are written in scripting languages (Perl, PHP), re-code them in compiled languages (C). 3) Use more powerful processors (faster clock, more cores, more processors). 4) Split the load across multiple hosts. This has the added advantage of not putting all your eggs in one basket -- you can take a host out of service for maintenance or upgrades. 5) If you are swapping, more RAM may help. Don't forget the fancy Digium codec translator card thingy! j -- _ -- 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] # as dial key - chan_dahdi
Hi, Can I set up '#' as dial key using the extensions fxs? I use chan_dahdi, and a TDM400P card. I'm testing and, nothing happens when I press #. thanks. -- Marcus Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.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] # as dial key - chan_dahdi
Asterisk reserves the # key. You can change this, but it is a buyer beware compile. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marcus Vinicius Sent: Tuesday, February 02, 2010 4:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] # as dial key - chan_dahdi Hi, Can I set up '#' as dial key using the extensions fxs? I use chan_dahdi, and a TDM400P card. I'm testing and, nothing happens when I press #. thanks. -- Marcus _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ 10 - Celebridades http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ - Música http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ - Esportes http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ -- _ -- 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 1.6.0.22, 1.6.1.14, and 1.6.2.2 Released
The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.6.0.22 * 1.6.1.14 * 1.6.2.2 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001. The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well. For more information about the details of this vulnerability, please read the security advisory AST-2009-009, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.22 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.14 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.2 Security advisory AST-2010-001 is available at: http://downloads.asterisk.org/pub/security/AST-2010-001.pdf Thank you for your continued support of Asterisk! -- _ -- 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] AST-2010-001: T.38 Remote Crash Vulnerability
Asterisk Project Security Advisory - AST-2010-001 ++ | Product| Asterisk| |--+-| | Summary| T.38 Remote Crash Vulnerability | |--+-| | Nature of Advisory | Denial of Service | |--+-| |Susceptibility| Remote unauthenticated sessions | |--+-| | Severity | Critical| |--+-| |Exploits Known| No | |--+-| | Reported On | 12/03/09| |--+-| | Reported By | issues.asterisk.org users bklang and elsto | |--+-| | Posted On | 02/03/10| |--+-| | Last Updated On| February 2, 2010| |--+-| | Advisory Contact | David Vossel dvossel AT digium DOT com | |--+-| | CVE Name | CVE-2010-0441 | ++ ++ | Description | An attacker attempting to negotiate T.38 over SIP can| | | remotely crash Asterisk by modifying the FaxMaxDatagram | | | field of the SDP to contain either a negative or | | | exceptionally large value. The same crash occurs when| | | the FaxMaxDatagram field is omitted from the SDP as | | | well.| ++ ++ | Resolution | Upgrade to one of the versions of Asterisk listed in the | || Corrected In section, or apply a patch specified in the | || Patches section.| ++ ++ | Affected Versions| || | Product | Release Series || |--++| | Asterisk Open Source | 1.6.x | All versions | |--++| |Asterisk Business Edition | C.3 | All versions | ++ ++ | Corrected In | || | Product | Release | |--+-| | Asterisk Open Source | 1.6.0.22 | |--+-| | Asterisk Open Source | 1.6.1.14 | |--+-| | Asterisk Open Source | 1.6.2.2 | |--+-| | | C.3.3.2 | ++ +-+ | Patches | |-| |
[asterisk-users] Queue problem, ringing agents.
I'm running Asterisk 1.6.0.21 and Aastra 57i phones. I'm having an issue with the agent phones ringing when someone is in our queue. The first phone will ring 3 to 4 times then the call will roll over to the next phone as expected. However, any phone after the first one will only ring once and the wait between that phone and the next will be as if it if ringing 3 to 4 times. My queues.conf is as follows: [general] ; ; Global settings for call queues ; (none exist currently) ; ; Note that a timeout to fail out of a queue may be passed as part of application call ; from extensions.conf: ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; example: Queue(dave|t|||45) #include queues_general_additional.conf #include queues_custom_general.conf [default] ; ; Default settings for queues (currently unused) ; #include queues_custom.conf #include queues_additional.conf #include queues_post_custom.conf queues_general_additional.conf: ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; persistentmembers=yes queues_custom_general.conf: is empty queues_custom.conf: is empty queues_additional.conf ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [1001] announce-frequency=45 announce-holdtime=yes announce-position=yes autofill=yes eventmemberstatus=no eventwhencalled=no joinempty=yes leavewhenempty=no maxlen=0 music=default periodic-announce-frequency=0 queue-callswaiting=queue-callswaiting queue-thankyou=queue-thankyou queue-thereare=queue-thereare queue-youarenext=queue-youarenext retry=20 ringinuse=yes strategy=leastrecent timeout=20 weight=0 wrapuptime=0 queues_post_custom.conf: is empty Any help will be greatly appreciated, Jeremy -- _ -- 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] MATH
I want to allow users to dial my DID
Then, hear my ginger3 intro
Then, depending on the number that they press, provide a total via MATH.
Comments. Will this work?
exten = 866,1,Goto(tommath,s,1)
[tommath]
exten = s,1,Read(NUMBER,ginger3,2,skip,5)
exten = s,n,Gotoif($[${NUMBER} = 14]?onefour)
exten = s,n,Gotoif($[${NUMBER} = 24]?twofour)
exten = s,n,Gotoif($[${NUMBER} = 34]?threefour)
exten = s,n,Gotoif($[${NUMBER} = 20]?done)
exten = s,playback(system) - error message
exten = s,n,Set(TOTAL=0)
exten = s,n(onefour),Set(TOTAL1=${MATH(${TOTAL}+500,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(twofour),Set(TOTAL2=${MATH(${TOTAL+TOTAL1}+200,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(threefour),Set(TOTAL3=${MATH(${TOTAL+TOTAL1+TOTAL2}+300,int)})
exten = s,n,Goto(tommath,s,1)
exten = s,n(done),SayNumber(${TOTAL=TOTAL1+TOTAL2+TOTAL3})
exten = s,n,playback(vm-goodbye)
exten = s,n,hangup
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[asterisk-users] asterisk video support and IPTV
Has anyone played with the idea of Asterisk as an H.264 multicast tool? I am wondering what the possibility would be to have some kind of machine with a capture card call asterisk over SIP and have asterisk make another hundred calls to subscribers. Then any H.264 compatible device (Android? Set top boxes? Plugin to MythTV?) would be able to receive a video/audio stream. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial multiple extensions and know who picks up call
Any updates on this? It looks like I can't update CDR(userfield) from
inside such a macro and have it written to the cdr record.
[macro-pstn-trigger]
exten = s,1,noop()
;exten = s,n,DumpChan()
exten = s,n,verbose(${DIALEDPEERNUMBER})
exten = s,n,verbose(cdr userfield ${CDR(userfield)})
exten = s,n,verbose(${CHANNEL(channeltype)} ${CHANNEL(peername)}
${CALLERID(num)})
exten = s,n,Set(CDR(userfield)=DIALEDPEERNUMBER=${DIALEDPEERNUMBER})
exten = s,n,verbose(cdr userfield ${CDR(userfield)})
the change shows up on the last line.
I might have to write to a file inside my macro?
On Wed, Sep 9, 2009 at 11:14 PM, Patrick asterisk-us...@ict-synergy.be wrote:
Thank you Jim, I'll check what I can find from the DumpChan() and keep
the mailing list posted.
Best regards,
Patrick
On Wed, Sep 9, 2009 at 16:57, Jim Dickenson dicken...@cfmc.com wrote:
Depending on version you are using you could use the M option on the
Dial command. I use 1.6.0.x and it works there. This causes a macro to
be executed when someone answers the call. There you have access to
many channel variables. Use DumpChan() to see what is available and go
from there.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 9, 2009, at 7:29 AM, Patrick wrote:
Thanks a lot Danny but isn't it another production proof way of
doing this ?
I've given a very simple example but in production, I can call up to
10 destinations simutaneously and there could have more than 30
concurrent calls, meaning that this solution doesn't sounds very
reliable in my case.
Is there any other way to retrieve the information ? Channel
variable ?
What a CDR(dst) returns after the Dial ? All destination or only the
destination that has answered ?
Thanks in advance
Patrick
On Wed, Sep 9, 2009 at 16:15, Danny Nicholasda...@debsinc.com wrote:
You could strip it out of a core show channels command. When 100
picks up
the call, the command will show an active call on 100. Of course
this
wouldn't be accurate if 101 also was on another call. You could do
an AGI
to start before the Dial and it could monitor and return the pickup
extension using AMI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick
Sent: Wednesday, September 09, 2009 9:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial multiple extensions and know who
picks up
call
Dear,
I'm currently using a Dial command with multiple destinations and
channels
eg: Dial(SIP/100SIP/101)
I simply would like to know, in real time during the call (from dial
plan or AGI), who has picked up the call.
Can I find this information in a variable somewhere ?
Thank you for your help
Patrick
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Re: [asterisk-users] MATH
On Tue, 2 Feb 2010, Thomas Perron wrote: I want to allow users to dial my DID Then, hear my ginger3 intro Then, depending on the number that they press, provide a total via MATH. Comments. Will this work? [snip] You've been asking this and related questions for days. Wouldn't it be faster to try it yourself? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 1.6.2 ?
This is usually due to an error with the SIP stack not being loaded due to an error - make sure that full logging is on and check your log file and search for ERROR and see if there is any mention to SIP (chan_sip.o etc), alternatively, start asterisk from the command like with asterisk -vdc and watch the output to screen for any errors at startup. Fix the error and SIP will start up. -- _ -- 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] CDR / billsec / originate / local chan
Hi All,
I have been running a environment with asterisk 1.4.20.1 for some time
now with no issue but have recently added some extra functionality
(enabled call recording via MixMonitor) and ran into some deadlock
issues which seem to be well documented with earlier 1.4.x releases so
have decided to take the plunge and upgrade. I decided to start testing
with 1.6.2 but have run into a couple of issues.
We run an environment that triggers outbound calls via AMI / asterisk
java and places the call upon answer back into a context that has IVR,
TTS/ VXML etc. Running with 1.4.20.1, we have not had any issues and
have been able to access all of the CDR fields in the h exten (using
endbeforehexten=yes in cdr.conf) however after upgrading all our post
answer related cdr information is reporting 0. Fields such as billsec,
answer etc all return either 0 or null which is causing a lot of issues
for us as we use some AGI post call via the h exten to perform
processing based on billable duration etc.
I have found a number of threads / articles etc discussing various
billsec related issues but it is hard to get a picture of what should
work on what version of asterisk. For example, I know that my
environment works on 1.4.20 but it is broken in 1.4.23 and seems to be
broken in 1.6.2 (pretty sure I tried 1.6.0.9 as well with the same result)
How t works in 1.4.20.1 is as follows:
We trigger call via Originate action as follows:
action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
context:.campaignType_5
priority:.1
channel:.SIP/trunk1/61212142321
And the campaignType_5 context looks similar to:
[campaignType_5_]
exten = s,1,Answer()
exten = s,n,Set(timestarted=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = s,n,Set(CALLSTATUS=0)
exten = s,n,Background(lyrics-louie-louie)
exten = s,n,WaitExten(5)
exten = s,n,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = s,n,Set(DIGITPRESSED=99)
exten = s,n,Set(TIMETOPRESS=${timestarted)})
exten = s,n,NoOp(Billsec is: ${CDR(billsec)})
exten = s,n,Hangup
exten = 1,1,Set(DIGITPRESSED=${EXTEN})
exten = 1,n,Set(TIMETOPRESS=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = 1,3,Playback(wait-moment)
exten = 1,4,Dial(Local/${calllogid}_${agentnumb...@campaigntype_5_agent/n)
exten = 1,5,Hangup
exten = h,1,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = h,2,GotoIf($[${TIMETOPRESS}foo = foo]?h,20)
exten = h,3,GotoIf($[${DIGITPRESSED}foo = foo]?h,10)
exten = h,4,Set(carrier=9)
exten =
h,n,AGI(agi://${DB(APPS/AGISERVER)}/ccoAgentActivityAgi.agi?BILLABLEDURATION=${CDR(billsec)}CALLLOGID=${CALLLOGID}CALLSTATUS=${CALLSTATUS}CAMPAIGNID=${CAMPAIGNID}DIGITPRESSED=${DIGITPRESSED}DURATION=${CDR(duration)}TARGETID=${TARGETID}TIMEFINISHED=${TIMEFINISHED}TIMEPRESSED=${TIMETOPRESS}TIME
STARTED=${TIMESTARTED}STATECHANGE=CALLDOWNNODEID=${NODEID})
exten = h,10,Set(DIGITPRESSED=77)
exten = h,11,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = h,12,Goto(h,3)
exten = h,20,Set(TIMETOPRESS=${timestarted})
exten = h,21,Goto(h,3)
exten = failed,1,Set(DIGITPRESSED=98)
exten = failed,2,Set(TIMETOPRESS=${timestarted})
exten = failed,3,Set(CALLSTATUS=6)
exten = failed,4,Set(timestarted=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = failed,5,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
As mentioned, in 1.4.20 this works fine with all cdr fields being
reported correctly, but on 1.6.2 (and various other versions, on the h
exten with the AGI call we get duration reported correctly but billsec = 0
We are not using Local channels here unless Asterisk is optimising the
outbound leg of the call into one invisibly.
On 1.6.2 I have also tried using a local channel for the outbound leg
with the originate looking like the following:
action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
context:.campaignType_5
priority:.1
channel:.Local/61212142...@outboundsip/n
And the Local context as follows;
[outboundsip]
exten = _XX.,1,Dial(SIP/trunk1/${EXTEN})
exten = _XX.,n,Hangup
exten = h,1,NoOp(Billsec is: ${CDR(billsec)})
In this configuration, whilst the outbound call goes out and billsec
gets reported correctly in the h exten, the call does not get bridged
back into the campaignType_5 context so none of the call processing
occurs. I cannot see any options that can be passed to the dial command
that may affect the bridging of the call back into the campaignType_5
context???
Does anyone out there know which way I need to hang my tongue out of my
mouth, how much i need to squint my eye and how far I need to tilt my
head (and to which side) to get 1.6.2 or any other post 1.4.20 version
of asterisk to report billsec etc back at the h exten for a call
established via an Originate action similar
Re: [asterisk-users] MATH
hi Steve, I am trying it and I am using the feedback from the group. In my view, that is the purpose; try, test, talk. Thanks for your interest. On Tue, Feb 2, 2010 at 7:15 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 2 Feb 2010, Thomas Perron wrote: I want to allow users to dial my DID Then, hear my ginger3 intro Then, depending on the number that they press, provide a total via MATH. Comments. Will this work? [snip] You've been asking this and related questions for days. Wouldn't it be faster to try it yourself? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] MATH
- Thomas Perron thomas.per...@gmail.com wrote: hi Steve, I am trying it and I am using the feedback from the group. In my view, that is the purpose; try, test, talk. Thanks for your interest. On Tue, Feb 2, 2010 at 7:15 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 2 Feb 2010, Thomas Perron wrote: I want to allow users to dial my DID Then, hear my ginger3 intro Then, depending on the number that they press, provide a total via MATH. Comments. Will this work? [snip] You've been asking this and related questions for days. Wouldn't it be faster to try it yourself? So what was the outcome when you tested that dial plan extension ? -- Thanks, Phil -- _ -- 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 realtime md5secret
Just remember, that after reload you will lose all registrations. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: 2010 m. vasario 2 d. 22:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip realtime md5secret On Tue, 2010-02-02 at 21:20 +0200, Emre Kurnaz wrote: Hi all, Does asterisk cache realtime sip md5secret values? I create a user over a web site and set a password as asd and I can login with that password. After a while I change my password and set it as 123. Although the password is set as 123 in the mysql database (I double checked), i can not login using the password 123, but with asd. So, am i missing a point? or is this how asterisk works? and Should I reload asterisk after adding a peer in the database? Any help would be appreciated. If you have rtcachefriends=yes set in your sip.conf file then you either have to wait until the peer expires or you have to reload sip so the peer is re read from the database. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] uri tel: instead of sip:accepted ?
2 feb 2010 kl. 11.20 skrev BERGANZ Francois: Hello all, Does asterisk accept uri tel: instead of sip: ? No, but I think it would be a good addition. /O -- _ -- 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] uri tel: instead of sip:accepted ?
On 02/03/2010 02:03 AM, Olle E. Johansson wrote: 2 feb 2010 kl. 11.20 skrev BERGANZ Francois: Hello all, Does asterisk accept uri tel: instead of sip: ? No, but I think it would be a good addition. Why? Just curious. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.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] Intel Atom based Asterisk server?
I currently have some Asterisk home servers on general pc hardware as well as a mission critical server asterisk pbx running on a Dell 2850 To reduce noise and power consumption i would like to migrate them all to an Intel Atom based solution, showstoppers so far were single NIC and single PCI slot motherboards. I found that Supermicro makes a Dual NIC board with one PCI slot and 2 PCI-Express slots (X7SLA-L) Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I am aware about other solutions but i do use the servers for some other tasks therefore don't want to move to a dedicated pbx box based on Soekris or the likes. Thanks for any input! -- _ -- 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] Intel Atom based Asterisk server?
- Remco Barendse aster...@barendse.to wrote: I currently have some Asterisk home servers on general pc hardware as well as a mission critical server asterisk pbx running on a Dell 2850 To reduce noise and power consumption i would like to migrate them all to an Intel Atom based solution, showstoppers so far were single NIC and single PCI slot motherboards. I found that Supermicro makes a Dual NIC board with one PCI slot and 2 PCI-Express slots (X7SLA-L) Has anyone tried running Asterisk + CentOS 5 on this (or any other) Atom board? Is the Atom platform able to handle the load of all the interrupts a TE110P or TDM400P card will generate ? I am aware about other solutions but i do use the servers for some other tasks therefore don't want to move to a dedicated pbx box based on Soekris or the likes. Thanks for any input! I am running a Atom Jetway JNC92 1.6GHz Dual Core Atom Motherboard with 2 x Gigabit LAN, 2GB RAM, TDM400P and it works great :) -- Thanks, Phil -- _ -- 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] uri tel: instead of sip:accepted ?
Where are these urls being input into asterisk? On Tue, Feb 2, 2010 at 11:11 PM, Alex Balashov abalas...@evaristesys.com wrote: On 02/03/2010 02:03 AM, Olle E. Johansson wrote: 2 feb 2010 kl. 11.20 skrev BERGANZ Francois: Hello all, Does asterisk accept uri tel: instead of sip: ? No, but I think it would be a good addition. Why? Just curious. -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.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 -- _ -- 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
