[asterisk-users] Asterisk, Hylafax and t38modem working together ?
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 ? Regards -- _ -- 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, Hylafax and t38modem working together ?
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 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
[asterisk-users] How to avoid automatic answer with callwaiting=yes on fxs channels?
I will make an example: A is an fxs phone with callwaiting=yes in chan_dahdi.conf X calls A. A answers. Y calls A. A hears the call waiting tone. Now if A hangs up before X, then A rings again (which is what I want). BUT if X hangs up first, then A automatically answers Y without even ringing. Is there a way to avoid it? If X hangs up first I want A to hear the busy tone until it hangs up too. Then I want A to ring again. Otherwise if both A and X hang up at the same time there is no way to know what happened. 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] Asterisk, Hylafax and t38modem working together ?
2012/10/8 Michel Verbraak 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 --- BeroFixhttp://www.beronet.com/product/berofix-gateways/--- ISDN32 No Asterisk in this case but it does work excelent. With the YaJHFC software you get a Windows/Linux/OSX printer driver. Very interesting to be aware of that : so many years of asterisk let me forgot hylafax could be used without asterisk at all. 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
Re: [asterisk-users] Asterisk, Hylafax and t38modem working together ?
2012/10/8 Michel Verbraak 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 --- BeroFixhttp://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. 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
Re: [asterisk-users] Fax for Asterisk success rates?
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 -- _ -- 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 registration Asterisk 1.8
Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register = 808:passw...@as2.x.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99 callerid=child2 808 disallow=all allow=ulaw allow=alaw username=808 secret=x dtmfmode=rfc2833 host=dynamic mailbox=808 nat=yes qualify=yes canreinvite=no == Extension Changed 800[sip-phones] new state Idle for Notify User 812 [Oct 8 09:48:37] NOTICE[12030]: chan_sip.c:26141 sip_poke_noanswer: Peer '808' is now UNREACHABLE! Last qualify: 1 == Using SIP RTP CoS mark 5 - Executing [808@sip-phones:1] Dial(SIP/815-00d8, SIP/808,20,t) in new stack [Oct 8 09:49:02] WARNING[12277]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/815-00d8' status is 'CHANUNAVA Any ideas? Thanks in Advance! -- _ -- 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] Sip registration Asterisk 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz Sent: Monday, October 08, 2012 12:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sip registration Asterisk 1.8 Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register = 808:passw...@as2.x.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99 callerid=child2 808 disallow=all allow=ulaw allow=alaw username=808 secret=x dtmfmode=rfc2833 host=dynamic mailbox=808 nat=yes qualify=yes canreinvite=no == Extension Changed 800[sip-phones] new state Idle for Notify User 812 [Oct 8 09:48:37] NOTICE[12030]: chan_sip.c:26141 sip_poke_noanswer: Peer '808' is now UNREACHABLE! Last qualify: 1 == Using SIP RTP CoS mark 5 - Executing [808@sip-phones:1] Dial(SIP/815-00d8, SIP/808,20,t) in new stack [Oct 8 09:49:02] WARNING[12277]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/815-00d8' status is 'CHANUNAVA Any ideas? Thanks in Advance! -- IIRC qualify=yes means you get 60 seconds; try it with qualify=300. -- _ -- 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 1.8.17.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix channel reference leak in ChanSpy. (Closes issue ASTERISK-19461. Reported by Irontec) * --- dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms (Closes issue ASTERISK-19610. Reported by Jean-Philippe Lord) * --- Fix bug where final queue member would not be removed from memory. (Closes issue ASTERISK-19793. Reported by Marcus Haas) * --- Fix memory leak when CEL is successfully written to PostgreSQL database (Closes issue ASTERISK-19991. Reported by Etienne Lessard) * --- Fix DUNDi message routing bug when neighboring peer is unreachable (Closes issue ASTERISK-19309. Reported by Peter Racz) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.17.0 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
[asterisk-users] Asterisk 10.9.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix channel reference leak in ChanSpy. (Closes issue ASTERISK-19461. Reported by Irontec) * --- dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms (Closes issue ASTERISK-19610. Reported by Jean-Philippe Lord) * --- Fix bug where final queue member would not be removed from memory. (Closes issue ASTERISK-19793. Reported by Marcus Haas) * --- Fix memory leak when CEL is successfully written to PostgreSQL database (Closes issue ASTERISK-19991. Reported by Etienne Lessard) * --- Fix DUNDi message routing bug when neighboring peer is unreachable (Closes issue ASTERISK-19309. Reported by Peter Racz) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.9.0 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
[asterisk-users] Asterisk 11.0.0-rc1 Now Available!
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! -- _ -- 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] 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? (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? -- Mitch -- _ -- 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!
One suggestion I have: Would it be helpful to know the revision number of rc1 in the release notes? I'm on a patched version of Asterisk from Doubango to deal with Chrome's non-standard ICE candidates, and unless this is included in rc1 (meaning rc1 is newer than what I have and also deals with Chrome's ICE issues) then I probably wouldn't upgrade. Also, I would prefer to check out from source but don't know the revision number to use. If I'm the only one that would benefit from this, then no worries, I'll deal with it. But if others would benefit from seeing a revision number/checking out from SVN, then maybe consider adding this to the release notes. :) Hope this helps! James On Mon, Oct 8, 2012 at 10:15 AM, Asterisk Development Team asteriskt...@digium.com 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.
Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
James Mortensen wrote: One suggestion I have: Would it be helpful to know the revision number of rc1 in the release notes? I'm on a patched version of Asterisk from Doubango to deal with Chrome's non-standard ICE candidates, and unless this is included in rc1 (meaning rc1 is newer than what I have and also deals with Chrome's ICE issues) then I probably wouldn't upgrade. Also, I would prefer to check out from source but don't know the revision number to use. Chrome Canary is actually using ICE according to the RFC these days and works fine with unpatched Asterisk. We don't presently include VP8 passthrough support though, so no video. If I'm the only one that would benefit from this, then no worries, I'll deal with it. But if others would benefit from seeing a revision number/checking out from SVN, then maybe consider adding this to the release notes. :) -- 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
[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? -- 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
[asterisk-users] blocking incoming call - asterisk 1.8
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-users
