Re: [asterisk-users] Asterisk, Hylafax and t38modem working together ?
Op 08-10-12 15:17, Olivier schreef: 2012/10/8 Michel Verbraak mic...@verbraak.org mailto:mic...@verbraak.org Op 08-10-12 09:24, Olivier schreef: Hi, I've read this thread in this list history http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657 Has anyone been successful when integrating latest version of Asterisk (10 or 1.8, for instance) with t38modem ? My target setup is: fax ---PSTN-- SPA3102 ---T38-- Asterisk with t38modem and Hylafax Suggestions ? Yup, YaJHFC http://www.yajhfc.de/ --- Hylafax --- t38modem --- BeroFix http://www.beronet.com/product/berofix-gateways/ --- ISDN32 By the way, which t38modem did you use ? On my debian system, version 1.2 is packaged and I wonder if it's worth the effort to use lastest 2.0 version. We use the 1.2.0-1 version on a debian system. No Asterisk in this case but it does work excelent. With the YaJHFC software you get a Windows/Linux/OSX printer driver. The BeroFix could be replaced with Asterisk but I do not have tested this. Regards, Michel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Unanswered calls
Hi, Any body has an idea on this ? I believe the configuration is correct. Is there any bug in this version ? Is there any version in 1.8 branch which has it working ? Please help. Regards Shanavaz. --- On Sat, 10/6/12, Shanavaz E A shanava...@yahoo.com wrote: From: Shanavaz E A shanava...@yahoo.com Subject: Re: [asterisk-users] CDR Unanswered calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, October 6, 2012, 11:28 AM Yes, Please see the following example. In version 1.4 of asterisk, we used to get atleast 2 records in the CDR table for one incoming call. One is the main record and second one is the record with the status of that particular extension number which answered the call. Additionally if any more extensions in the queue was tried, and if it was failed or busy or not answered, an additional record for each of those extension will be created in the CDR table. For example: | calldate | src | dst | dcontext | channel | dstchannel | lastapp | lastdata | duration | billsec | disposition | | 2012-09-01 20:02:54 | 9123456733 | s | queue-cbkn | DAHDI/7-1 | Agent/3009 | Queue | BookingQ|tT|||30|myagi1.agi | 156 | 156 | ANSWERED | | 2012-09-01 20:03:02 | 9123456733 | 305 | from-internal | Local/305@from-internal-df2a,2 | SIP/305-00087978 | Dial | SIP/305||tT | 0 | 0 | BUSY | | 2012-09-01 20:03:02 | 9123456733 | 307 | from-internal | Local/307@from-internal-dd81,2 | SIP/307-00087979 | Dial | SIP/307||tT | 5 | 0 | ANSWERED | Here the first record is the main record. Second one is extn 305 was tried but it was BUSY and third record is extn 307 was tried and it answered that call. So here totally three records were created for a single call. But in asterisk 1.8 only the first record is being created. I need all the records for all extensions which was tried in the queue. I hope you got the point Regards Shanavaz. --- On Sat, 10/6/12, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] CDR Unanswered calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, October 6, 2012, 1:52 AM On Fri, Oct 5, 2012 at 4:51 AM, Shanavaz E A shanava...@yahoo.com wrote: Hi, No replies until now. Some one please help... There must be some people who are using it... Thanks Can you provide an example of what you expect it to be doing (from the old version) and what it is doing now (from the new version)? I'm talking examples of the table rows in question. Is it recording the call, just labeling it answered instead of unanswered? I've never seen asterisk simply not record a call in whatever CDR backend you're using, regardless of disposition. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sends wrong fxs 'Idle' hints
Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] blocking incoming call - asterisk 1.8
Hello Yes, has a berckeley database, wirh function blackllist Regards On Oct 9, 2012 12:51 AM, Joseph syscon...@gmail.com wrote: Can someone refresh my memory how blocking incoming call works based on caller ID in Asterisk 1.8? If I remember correctly in asterisk 1.4 it was possible to block caller ID from the command line, asterisk had some internal database I think. -- Joseph -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/09/2012 12:28 AM, Brett Lehrer wrote: How many fax and voice calls (which codecs for tha latter ones ?) are on average using your DSL line ? 1. Previously, I experienced failures during the process of converting incoming PDF documents into ready-to-send fax image files while the reverse process (from a fax file into a PDF or whatever document) never failed. I would be curious to check if a greater failure rate for outbound faxing (greater than inbound faxing failure rate) could simply comes from image processing, before any transmission. 2. Though your DSL line may have enough bandwidth from your location to its DSLAM, chances are packets are dropped or delivered too late for T.38 faxing. An interesting test would be to use an Asterisk PBX hosted somewhere at close range from netVortex fax gateways : that would remove most networking issues out of the equation. I'll have to look more closely into what codecs we traditionally use, but g.722 up and ulaw down is common. Generally don't have more than 2-3 calls active at once. At most, 5, and that's a rarity. Record for fax is 4 simultaneous send/receive, but typically just 1, maybe 2. I imagine that's encroaching on the upper limits of the 768 kbps upspeed. I've wondered about how lag might impact the problem but I just don't know how I'd go about testing it properly without spending a bunch of money on hosting. I do my PDF - TIFF conversion on another machine with ghostscript. Here's the line: gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg4 -sOutputFile=TIFF_FILENAME -f PDF_FILENAME I changed from tiffg3 to tiffg4 because the filesize got cut in half assuming that the less time spent transmitting, the less chance there was to run into a problem that might stop the fax. However, most failures that I've looked at seem to occur immediately or fail to connect at all, rather than get cut off due to a hiccup in the connection. Brett Lehrer A FAX can only be sent in ECM mode when using tiffg4 format. It will have to be recoded into tiffg3 format if ECM is inhibited, which it far too often is. On the other hand, if you are using ECM any decent FAX system (e.g. spandsp) will recode into tiffg4, and really good ones (e.g. the very latest spandsp) may recoed into T.85/JBIG, for faster transmission times. Digium don't seem to specify what FFA does in this area. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP Driver and VoiceMail
- Original Message - From: Patrick Lists asterisk-l...@puzzled.xs4all.nl To: asterisk-users@lists.digium.com Sent: Friday, 5 October, 2012 11:46:48 AM Subject: Re: [asterisk-users] LDAP Driver and VoiceMail On 10/04/2012 10:00 PM, Phil Daws wrote: Hello: I am investigating the possibility of using LDAP for storing certain Asterisk configuration parameters. I have examined res_ldap.conf and see where mailbox can be defined from AstAccountMailbox but I do not see where the password can be stored ? I've never looked at res_ldap but wouldn't a look at the schema tell you that? Regards, Patrick -- Have successfully converted the OpenLDAP schema to a 389 DS version and imported it into my DIT. Am now looking at https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver so see how to configure Asterisk to query LDAP for the information. I see that one needs to configure res_ldap.conf to set up the necessary server and bind variables for authentication followed by extconfig.conf. What I am unsure of though is the following syntax: voicemail = ldap,ou=voicemail,dc=example,dc=domain,voicemail so this would tell the voicemail module to query LDAP for the respective data under the OU voicemail. Now what if the query failed ? would it then look to voicemail.conf so that you can have a mix between LDAP and flat file configuration files ? All help appreciated :) Regards, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING T.30 ECM carrier not found
I'm working on setting up incoming fax reception on our * server. The majority of faxes come through fine. However each timed a fax comes in, I get a bunch of this: WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found Should this be of concern to me? A snip of the log is below. Kind Regards, Chris -- Executing [19108929322@from-trunk:6] Set(SIP/foobar_trunk_did_b-0174, CALLERPRES()=allowed_not_screened) in new stack -- Executing [19108929322@from-trunk:7] Set(SIP/foobar_trunk_did_b-0174, FAX_DEST=ext-fax^166^1) in new stack -- Executing [19108929322@from-trunk:8] Answer(SIP/foobar_trunk_did_b-0174, ) in new stack -- Executing [19108929322@from-trunk:9] Wait(SIP/foobar_trunk_did_b-0174, 4) in new stack == Redirecting 'SIP/foobar_trunk_did_b-0174' to fax extension due to CNG detection == Spawn extension (from-trunk, fax, 1) exited non-zero on 'SIP/foobar_trunk_did_b-0174' -- Executing [fax@from-trunk:1] Goto(SIP/foobar_trunk_did_b-0174, ext-fax,166,1) in new stack -- Goto (ext-fax,166,1) -- Executing [166@ext-fax:1] Set(SIP/foobar_trunk_did_b-0174, FAX_FOR=Fax (166)) in new stack -- Executing [166@ext-fax:2] NoOp(SIP/foobar_trunk_did_b-0174, Receiving Fax for: Fax (166), From: +18009806858 +18009806858) in new stack -- Executing [166@ext-fax:3] Set(SIP/foobar_trunk_did_b-0174, FAX_RX_EMAIL=f...@foobar.com) in new stack -- Executing [166@ext-fax:4] Goto(SIP/foobar_trunk_did_b-0174, s,receivefax) in new stack -- Goto (ext-fax,s,3) -- Executing [s@ext-fax:3] StopPlayTones(SIP/foobar_trunk_did_b-0174, ) in new stack -- Executing [s@ext-fax:4] ReceiveFAX(SIP/foobar_trunk_did_b-0174, /var/spool/asterisk/fax/1349791968.502.tif,f) in new stack -- Channel 'SIP/foobar_trunk_did_b-0174' receiving FAX '/var/spool/asterisk/fax/1349791968.502.tif' [2012-10-09 10:13:03] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:04] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:04] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:24] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:13:24] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:07] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:07] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:20] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found [2012-10-09 10:14:20] WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM carrier not found pbx1*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends wrong fxs 'Idle' hints
Il 09/10/2012 13:34, Niccolò Belli ha scritto: Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the Idle state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him because the light on their function keys switched off! Cheers, Niccolò I made a video of the bug: http://files.linuxsystems.it/files/dahdi_hints_bug.webm Can someone help me? Thanks, Niccolò -- http://www.linuxsystems.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
Asterisk 1.8 (a) We will have a group of 4 analog lines into a Digium card that will be used for local calls. What is the best way to use those lines as a pool for outbound calls? Can I use ChanIsAvail(), listing those 4 channels, and then use the first one returned? There are lots of things documented in chan_dahdi.conf.sample. The following option will assign channels 1-4 to group 1. ; Logical groups can be assigned to allow outgoing roll-over. Groups range ; from 0 to 63, and multiple groups can be specified. By default the ; channel is not a member of any group. ; ; Note that an explicit empty value for 'group' is invalid, and will not ; override a previous non-empty one. The same applies to callgroup and ; pickupgroup as well. ; group=1 channel = 1-4 Then you can dial from that group of channels: same = n,Dial(DAHDI/g1/5551212) /* * data is ---v * Dial(DAHDI/pseudo[/extension[/options]]) * Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/ispan[/extension[/options]]) * Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]]) * * i - ISDN span channel restriction. * Used by CC to ensure that the CC recall goes out the same span. * Also to make ISDN channel names dialable when the sequence number * is stripped off. (Used by DTMF attended transfer feature.) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * rcadance# - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ (b) For emergency calls, I want to be able to force one of these lines available if all are in use. Will SoftHangup() do that? If so, do I need to Wait() after a SoftHangup() before trying to use it? SoftHangup() should do what you want for this. You need to have a wait so the soft hangup will have a chance to be recognized. I would also suggest that if you use g1 in your normal dial, you should use the highest channel as your emergency line. That channel will be the last used by the group so an emergency call will be least likely to kick off an established call. Another approach is to attempt to dial the emergency call normally. If the first attempt fails, then kick an established call. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax_provision_version: ast_db_get failed
After upgrading to Asterisk 1.8.15.1 I'm constantly getting this error on the command line: ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get failed to retrieve iax/provisioning/cache Can somebody explain what it is and how to fix it? Since you say this happens repeatedly, Asterisk may not have the correct permissions to access the database. /var/lib/asterisk/astdb Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/09/2012 07:40 AM, Steve Underwood wrote: On 10/09/2012 12:28 AM, Brett Lehrer wrote: How many fax and voice calls (which codecs for tha latter ones ?) are on average using your DSL line ? 1. Previously, I experienced failures during the process of converting incoming PDF documents into ready-to-send fax image files while the reverse process (from a fax file into a PDF or whatever document) never failed. I would be curious to check if a greater failure rate for outbound faxing (greater than inbound faxing failure rate) could simply comes from image processing, before any transmission. 2. Though your DSL line may have enough bandwidth from your location to its DSLAM, chances are packets are dropped or delivered too late for T.38 faxing. An interesting test would be to use an Asterisk PBX hosted somewhere at close range from netVortex fax gateways : that would remove most networking issues out of the equation. I'll have to look more closely into what codecs we traditionally use, but g.722 up and ulaw down is common. Generally don't have more than 2-3 calls active at once. At most, 5, and that's a rarity. Record for fax is 4 simultaneous send/receive, but typically just 1, maybe 2. I imagine that's encroaching on the upper limits of the 768 kbps upspeed. I've wondered about how lag might impact the problem but I just don't know how I'd go about testing it properly without spending a bunch of money on hosting. I do my PDF - TIFF conversion on another machine with ghostscript. Here's the line: gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg4 -sOutputFile=TIFF_FILENAME -f PDF_FILENAME I changed from tiffg3 to tiffg4 because the filesize got cut in half assuming that the less time spent transmitting, the less chance there was to run into a problem that might stop the fax. However, most failures that I've looked at seem to occur immediately or fail to connect at all, rather than get cut off due to a hiccup in the connection. Brett Lehrer A FAX can only be sent in ECM mode when using tiffg4 format. It will have to be recoded into tiffg3 format if ECM is inhibited, which it far too often is. On the other hand, if you are using ECM any decent FAX system (e.g. spandsp) will recode into tiffg4, and really good ones (e.g. the very latest spandsp) may recoed into T.85/JBIG, for faster transmission times. Digium don't seem to specify what FFA does in this area. Steve A little puzzled. Do you mean: 1. tiffg4 encoded fax will(might?) fail if ECM is inhibited at either send or receive. 2. tiffg3 will work if ECM is inhibited. 3. If ECM is not inhibited, any decent fax system, will reencode tiffg3 to tiffg4. Therefore we should encode to tiffg3 and let spandsp determine if it should be rencoded to tiffg4 (or T.85/JBIG)? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax_provision_version: ast_db_get failed
On 10/09/12 10:55, Richard Mudgett wrote: After upgrading to Asterisk 1.8.15.1 I'm constantly getting this error on the command line: ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get failed to retrieve iax/provisioning/cache Can somebody explain what it is and how to fix it? Since you say this happens repeatedly, Asterisk may not have the correct permissions to access the database. /var/lib/asterisk/astdb Richard Yes, it happens repeatedly. I just check and the database is owned by asterisk: -rw-r--r-- 1 asterisk asterisk 8192 Oct 9 10:31 /var/lib/asterisk/astdb -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
Excellent. I'll give it a try. (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) Mitch On 10/09/2012 10:48 AM, Richard Mudgett wrote: There are lots of things documented in chan_dahdi.conf.sample. The following option will assign channels 1-4 to group 1. ; Logical groups can be assigned to allow outgoing roll-over. Groups range ; from 0 to 63, and multiple groups can be specified. By default the ; channel is not a member of any group. ; ; Note that an explicit empty value for 'group' is invalid, and will not ; override a previous non-empty one. The same applies to callgroup and ; pickupgroup as well. ; group=1 channel = 1-4 Then you can dial from that group of channels: same = n,Dial(DAHDI/g1/5551212) /* * data is ---v * Dial(DAHDI/pseudo[/extension[/options]]) * Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/ispan[/extension[/options]]) * Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]]) * * i - ISDN span channel restriction. * Used by CC to ensure that the CC recall goes out the same span. * Also to make ISDN channel names dialable when the sequence number * is stripped off. (Used by DTMF attended transfer feature.) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * rcadance# - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ (b) For emergency calls, I want to be able to force one of these lines available if all are in use. Will SoftHangup() do that? If so, do I need to Wait() after a SoftHangup() before trying to use it? SoftHangup() should do what you want for this. You need to have a wait so the soft hangup will have a chance to be recognized. I would also suggest that if you use g1 in your normal dial, you should use the highest channel as your emergency line. That channel will be the last used by the group so an emergency call will be least likely to kick off an established call. Another approach is to attempt to dial the emergency call normally. If the first attempt fails, then kick an established call. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote: (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) You can use dynamic DAHDI spans to simulate this on a single box if you would with DAHDI-Linux 2.6.0+. Something like: In /etc/dahdi/system.conf use: dynamic=loc,1:0,4,0 fxsks=49-52 dynamic=loc,1:1,4,0 fxoks=53-56 loadzone= us defaultzone = us And in /etc/asterisk/chan_dahdi.conf: signalling=fxs_ks context=pstn group=0 channel=1 channel=2 channel=3 channel=4 signalling=fxo_ks context=simulation group=63 channel=51 channel=52 channel=53 channel=54 Now you can start up asterisk and group 0 will be your normal group and you can answer these lines in the simulation context. Dynamic local spans have been around for awhile but I've only used them on a regular basis since 2.6.0+. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
Minor correction below: On Tue, Oct 09, 2012 at 12:32:44PM -0500, Shaun Ruffell wrote: On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote: (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) You can use dynamic DAHDI spans to simulate this on a single box if you would with DAHDI-Linux 2.6.0+. Something like: In /etc/dahdi/system.conf use: dynamic=loc,1:0,4,0 fxsks=49-52 Should make the above line: fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=53-56 loadzone= us defaultzone = us And in /etc/asterisk/chan_dahdi.conf: signalling=fxs_ks context=pstn group=0 channel=1 channel=2 channel=3 channel=4 signalling=fxo_ks context=simulation group=63 channel=51 channel=52 channel=53 channel=54 Now you can start up asterisk and group 0 will be your normal group and you can answer these lines in the simulation context. Dynamic local spans have been around for awhile but I've only used them on a regular basis since 2.6.0+. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.13 Now Available
The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. (Issue AST-598. Reported by Trey Blancher) * --- Implement handling a multi-channel RESTART request. (Closes issue PRI-93. Reported by Marcin Kowalczyk) * --- Removed MDL/TEI management configuration warning message. (Closes issue PRI-137. Reported by Bart Coninckx) * --- Allow passing compiler flags (CFLAGS, LDFLAGS) (Closes issue PRI-144. Reported by Tzafrir Cohen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.13 Now Available
On 10/09/2012 02:00 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. (Issue AST-598. Reported by Trey Blancher) * --- Implement handling a multi-channel RESTART request. (Closes issue PRI-93. Reported by Marcin Kowalczyk) * --- Removed MDL/TEI management configuration warning message. (Closes issue PRI-137. Reported by Bart Coninckx) * --- Allow passing compiler flags (CFLAGS, LDFLAGS) (Closes issue PRI-144. Reported by Tzafrir Cohen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13 Thank you for your continued support of Asterisk! So as you can tell, this is actually libpri, *not* Asterisk. The script responsible has been sternly reprimanded. Sorry for any confusion! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not able to post to list
Hi who is responsible for this mailing list? i am not able to post to it. Br Adnan Sent from my iPhone On 9 okt 2012, at 21:04, Matthew Jordan mjor...@digium.com wrote: On 10/09/2012 02:00 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. (Issue AST-598. Reported by Trey Blancher) * --- Implement handling a multi-channel RESTART request. (Closes issue PRI-93. Reported by Marcin Kowalczyk) * --- Removed MDL/TEI management configuration warning message. (Closes issue PRI-137. Reported by Bart Coninckx) * --- Allow passing compiler flags (CFLAGS, LDFLAGS) (Closes issue PRI-144. Reported by Tzafrir Cohen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13 Thank you for your continued support of Asterisk! So as you can tell, this is actually libpri, *not* Asterisk. The script responsible has been sternly reprimanded. Sorry for any confusion! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failover router recommendation
I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? TIA, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to post to list
On Tue, Oct 9, 2012 at 12:16 PM, Adnan 112linuxstockh...@gmail.com wrote: Hi who is responsible for this mailing list? i am not able to post to it. You just did. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
Edgewater 4350 or cheaper vigor 2910 dreytech On Tue, Oct 9, 2012 at 3:24 PM, Mike Diehl mdiehlena...@gmail.com wrote: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? TIA, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- robertros...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
I found that I had to chmod 666 /dev/dahdi/* to allow asterisk to use the simulation channels. The /dev/dahdi directory seems to be recreated when dahdi starts. Here is what I finally came up with that works for me. system.conf dynamic=loc,1:0,4,0 fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=5-8 loadzone= us defaultzone = us chan_dahdi.conf signalling=fxs_ks context=simulation group=0 channel=1 signalling=fxo_ks context=dummy group=63 channel=5 I can now dial out on group 63 and it rings in the simulation context, which I forward to a SIP phone for testing. Is that what you expected to see? Mitch On 10/09/2012 12:40 PM, Shaun Ruffell wrote: Minor correction below: On Tue, Oct 09, 2012 at 12:32:44PM -0500, Shaun Ruffell wrote: On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote: (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) You can use dynamic DAHDI spans to simulate this on a single box if you would with DAHDI-Linux 2.6.0+. Something like: In /etc/dahdi/system.conf use: dynamic=loc,1:0,4,0 fxsks=49-52 Should make the above line: fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=53-56 loadzone= us defaultzone = us And in /etc/asterisk/chan_dahdi.conf: signalling=fxs_ks context=pstn group=0 channel=1 channel=2 channel=3 channel=4 signalling=fxo_ks context=simulation group=63 channel=51 channel=52 channel=53 channel=54 Now you can start up asterisk and group 0 will be your normal group and you can answer these lines in the simulation context. Dynamic local spans have been around for awhile but I've only used them on a regular basis since 2.6.0+. Cheers, Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
On Tue, Oct 09, 2012 at 03:41:49PM -0500, Mitch Claborn wrote: I found that I had to chmod 666 /dev/dahdi/* to allow asterisk to use the simulation channels. The /dev/dahdi directory seems to be recreated when dahdi starts. Here is what I finally came up with that works for me. system.conf dynamic=loc,1:0,4,0 fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=5-8 loadzone= us defaultzone = us chan_dahdi.conf signalling=fxs_ks context=simulation group=0 channel=1 signalling=fxo_ks context=dummy group=63 channel=5 I can now dial out on group 63 and it rings in the simulation context, which I forward to a SIP phone for testing. Is that what you expected to see? Basically yes. That should allow you to try things out without access to the actual hardware and yet still use chan_dahdi. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
I am sure Mikrotik routers will do this also, although I have not tried it. Niccolò Belli darkba...@linuxsystems.it wrote: Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling out on a group of DAHDI lines
Here's what I came up with. Works find with the simulated DAHDI dynamic
local channels. I'll find out later in the week how it works with real
hardware.
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1) ;
exten =999,1,Goto(dialpsap,1)
exten =112,1,Goto(dialpsap,1)
exten =dialpsap,1,Verbose(1,Call initiated to PSAP!)
same =n(dialit),Dial(${LOCAL}/${EMERGENCY},30)
same =n,Verbose(2,DIALSTATUS=${DIALSTATUS})
same =n,GotoIf($[${DIALSTATUS} = ANSWER]?good)
same =n(hu),SoftHangup(${EMERGENCY_CHANNEL},a)
same =n,Wait(2)
same =n,Goto(dialit)
same =n(good),NoOp(call good)
same =n,Hangup()
Mitch
On 10/09/2012 10:48 AM, Richard Mudgett wrote:
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that
will
be used for local calls. What is the best way to use those lines as
a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
There are lots of things documented in chan_dahdi.conf.sample. The
following option will assign channels 1-4 to group 1.
; Logical groups can be assigned to allow outgoing roll-over. Groups range
; from 0 to 63, and multiple groups can be specified. By default the
; channel is not a member of any group.
;
; Note that an explicit empty value for 'group' is invalid, and will not
; override a previous non-empty one. The same applies to callgroup and
; pickupgroup as well.
;
group=1
channel = 1-4
Then you can dial from that group of channels:
same = n,Dial(DAHDI/g1/5551212)
/*
* data is ---v
* Dial(DAHDI/pseudo[/extension[/options]])
* Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]])
*
Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]])
* Dial(DAHDI/ispan[/extension[/options]])
*
Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]])
*
* i - ISDN span channel restriction.
* Used by CC to ensure that the CC recall goes out the same span.
* Also to make ISDN channel names dialable when the sequence number
* is stripped off. (Used by DTMF attended transfer feature.)
*
* g - channel group allocation search forward
* G - channel group allocation search backward
* r - channel group allocation round robin search forward
* R - channel group allocation round robin search backward
*
* c - Wait for DTMF digit to confirm answer
* rcadance# - Set distintive ring cadance number
* d - Force bearer capability for ISDN/SS7 call to digital.
*/
(b) For emergency calls, I want to be able to force one of these
lines
available if all are in use. Will SoftHangup() do that? If so, do I
need to Wait() after a SoftHangup() before trying to use it?
SoftHangup() should do what you want for this. You need to have a wait
so the soft hangup will have a chance to be recognized.
I would also suggest that if you use g1 in your normal dial, you should use the
highest channel as your emergency line. That channel will be the last used
by the group so an emergency call will be least likely to kick off an
established
call.
Another approach is to attempt to dial the emergency call normally. If the
first
attempt fails, then kick an established call.
Richard
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Re: [asterisk-users] Failover router recommendation
On 10/9/2012 3:52 PM, Niccolò Belli wrote: http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Do you have your phones set for a short register time? Otherwise the far end might have stale contact information to send incoming calls back to. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP redundancy
Hi, I am investigating about some SIP redundancy method. I found this article http://academiccommons.columbia.edu/download/fedora_content/download/ac:109760/CONTENT/cucs-011-04.pdf and I will try to implement. But, I'd like to ask you, somebody had implemented some method? Do you have experiences or sugestions about this? Thank you very much. Alonso -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
On 10/08/2012 05:15 PM, Asterisk Development Team wrote: The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 11.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases All interested users of Asterisk are encouraged to participate in the Asterisk 11 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Asterisk 11 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 A short list of new features includes: * A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle specification, and the original Google Talk protocol. * Support for the WebSocket transport for chan_sip. * SIP peers can now be configured to support negotiation of ICE candidates. * The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally. * Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan. * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the callee channels, but before any actions have been taken to actually dial the callee channels. * Log messages can now be easily associated with a certain call by looking at a new unique identifier, Call Id. Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call. * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules. * The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s). * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. * Support for DTLS-SRTP in chan_sip. * Support for named pickupgroups/callgroups, allowing any number of pickupgroups and callgroups to be defined for several channel drivers. * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation A full list of all new features can also be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/11/CHANGES For a full list of changes in the current release, please see the ChangeLog. http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1 Thank you for your continued support of Asterisk! Thanks for all the great work. We've started using the silk codec a lot for phone app voip. We've found it the most effective low bit rate (16K) codec. Could we get a release 11 version of the silk codec in http://downloads.digium.com/pub/telephony/codec_silk/ ? That way we could start messing with RC 1. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
Re: [asterisk-users] Failover router recommendation
Il 09.10.2012 23:04 James Sharp ha scritto: Do you have your phones set for a short register time? Otherwise the far end might have stale contact information to send incoming calls back to. Actually I use the failover only for the nat clients, my pbx has a public ip on the interface and it receives the incoming calls from PRI (which I use as outgoing fallback too). externaddr should be another thing you should take care of. Let me know how you will work around such things, my main focus had been nat clients and I did just a few tests with asterisk. Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
On 10/10/2012, at 9:54 AM, cov...@ccs.covici.com wrote: I am sure Mikrotik routers will do this also, although I have not tried it. Mikrotik can do this but it takes some setup. They are very powerful but what you are asking is complex and may require the following - 2 ethernet upstreams or the ability to use PPOE to access DSL line. You can use DSL modems and more NAT but its important the device recognises the line has gone down. Mikrotiks don't do PPOA so if the DSL link uses PPOA then you need to have more natting. - hard to use SIP externip as the externip changes depending on which link it comes from. The router can't influence this. - your failover router has to reset itself to use the alternate DNS, and this sometime is a problem, more so when one link comes back the system has to change back. Some Mikrotiks I have used have trouble doing this. - its easiest if you route everything out one link then if that fails route everything out the other link, if you just want some traffic rerouted you have to distinguish it and route it with a policy route which can get complex Its not straight forward. Although I believe there are devices out there that have it all set up for you. A much simpler way is to have asterisk have two separate trunks routed out two different links and then use asterisk to do any failover, then it solves other problems eg. the link is okay but the provider has gone down You can make calls out the secondary trunks, and put a call forward on the primary numbers to the secondary if there is a failure Cheers Duncan Niccolò Belli darkba...@linuxsystems.it wrote: Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so that if one goes down, he can use the other for phone service. So, I'd like to get a recommendation for a relatively inexpensive router that can perform this function. Also, when the failover occurs, the phone's IP address will obviously change. So, how can/should I configure this to minimize my customer's down-time? http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Cheers, Niccolò -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Regarding caller ID and security
Hi all, I am new to Asterisk, and would like to begin by saying that it is an absolutely fantastic system. Seems incredibly stable, well tested, and easy to use. Now, to my question. I am making a mix between a personal ads and a voicemail service, where I want each user to be able to submit an ad that others can respond to by recording messages that go into this users inbox. My original thought was to base this purely on the CALLERID(num) value, but quickly discovered that this is a bit unreliable. Sometimes when I would call in it'd say anonymous, other times it would give me a bunch of zero's, other times it would show me my real phone number, and once it actually gave me just random digits. I do have a wait call after answering but before my first soundf ile is triggered, in my pickup context. I am wondering what the best way to approach this is? Do I ask the user to enter their phone number, and then generate a code based upon this that will then serve as a password when you call back? Do I attempt to use CALLERID(num) to detect returning users, or is this not adviseable from a security perspective? Preferably, I would like to avoid using a code altogether but I am told that it is relatively easy to spoof phone numbers to hack into someone else's inbox. Note that I do not plan to allow direct SIP calls, only through a PSTN/SIP provider where the IP address is on a whitelist. Any tips on how to approach this would be highly appreciated. Basically I want to make it as easy as possible for my users, but maintain high security. Thanks in advance for any help, and thanks once again to the developers of Asterisk for making such an excellent tool! Kind regards, Philip Bennefall P.S. I also wanted to know whether there is a function to check if a string contains only digits? This would be useful as a sanity check before I look up the phone number in the MySql database, if I do decide to use CALLERID(num) in this way.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how does GOTO_ON_BLINDXFR work?
10.9.0. I'm trying to have a setup where hitting # sends the called party to the confbridge. I've set GOTO_ON_BLINDXFR: CLI dialplan show globals . GOTO_ON_BLINDXFR=tel-incoming^confbridge^1 (Also tried tel-incoming,confbridge,1 and using | ) but it doesn't work: Dial(DAHDI/1-1, DAHDI/4/xxxyyy,,tT) in new stack -- Called DAHDI/4/xxxyyy ... -- DAHDI/4-1 answered DAHDI/1-1 -- Started music on hold, class 'default', on DAHDI/4-1 -- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en') [Oct 9 18:10:33] WARNING[28164]: features.c:2367 builtin_blindtransfer: No digits dialed. -- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en') sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a=recvonly
I am setting up with meetme a conf with X number of asterisk boxes and other devices and phones. I am using the l parameter for all devices being listen only but I'm not sure thats happening as I am getting some feedback (some devices are close to each other like 5 feet). How do I ensure that a=recvonly is being set or sent when bringing a device into the meetme? Can I added that to SIPADDHEADER or something? THere is only one device talking and all others should just be listening. I am using 1.4.43 Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a=recvonly
Jerry Geis wrote: Hola, I am setting up with meetme a conf with X number of asterisk boxes and other devices and phones. I am using the l parameter for all devices being listen only but I'm not sure thats happening as I am getting some feedback (some devices are close to each other like 5 feet). I've checked the code in Meetme and confirmed that feature should be working as expected. I would suggest you do some additional testing to confirm it isn't. How do I ensure that a=recvonly is being set or sent when bringing a device into the meetme? Can I added that to SIPADDHEADER or something? Asterisk doesn't allow you to manipulate the SDP as such so I'm afraid there is no way for you to change that value. Internally within applications they just discard the media received under such circumstances. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
