[asterisk-users] http://downloads.asterisk.org/pub/telephony/dahdi-linux/ not showing lastest changelog
Hi, Is it me or unlike http://downloads.asterisk.org/pub/telephony/asterisk/, http://downloads.asterisk.org/pub/telephony/dahdi-linux/ is not showing lastest changelog anymore ? 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] http://downloads.asterisk.org/pub/telephony/dahdi-linux/ not showing lastest changelog
On Mon, Jul 14, 2014 at 2:56 AM, Olivier oza.4...@gmail.com wrote: Is it me or unlike http://downloads.asterisk.org/pub/telephony/asterisk/, http://downloads.asterisk.org/pub/telephony/dahdi-linux/ is not showing lastest changelog anymore ? dahdi project dropped generating this specific changelog during the svn to git migration. Honestly, we didn't think anybody used it and the information is replicated elsewhere. diffstats and patch shortlogs are generated for each release email. We also keep sane shortlog synopsis for each patch which makes it a lot easier to browse the git logs here, if you're looking for something specific: http://git.asterisk.org/gitweb/?p=dahdi/linux.git http://git.asterisk.org/gitweb/?p=dahdi/tools.git -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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] Call drop on Aastra SIP phones
Hello everybody, I'm having issues with calls being dropped on Aastra phones, when the call is on hold. Tested with models 6863i and 6867i. I've figured that the call is dropped by Asterisk when it reaches the rtpholdtimeout limit. I've reported the issue to Aastra, asking them to implement some kind of RTP keep-alive feature on their phones. Maybe the phone could send some RTCP frame (or an empty RTP frame) just to prove it is alive. Unfortunately Aastra said the hold behaviour on the phone is correct, as per RFC 3264, section 8.4, 4th paragraph: Typically, when a user presses hold, the agent will generate an offer with all streams in the SDP indicating a direction of sendonly, and it will also locally mute, so that no media is sent to the far end, and no media is played out. They can implement the RTP keep-alive feature only if there is some RFC describing that behaviour. Is Aastra correct? Should I configure Asterisk with rtpholdtimeout=0 to solve this issue and make Asterisk RFC compliant? Or should Asterisk rtpholdtimeout code take the phone RFC behaviour into account and don't drop the call? Best regards, -- Bruno Rocha -- _ -- 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] Issue with new Te133
Have a new TE133 in a dell R320. dahdi_cfg -vvv shows: DAHDI Tools Version - 2.9.1 DAHDI Version: 2.9.1 Echo Canceller(s): Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 24: D-channel (Default) (Echo Canceler: none) (Slaves: 24) 7 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Not sure why??? system.conf shows: cat system.conf loadzone=us defaultzone=us span=1,0,0,esf,b8zs bchan=1-6 dchan=24 echocanceller=mg2,1-6 I installed from source. I am running asterisk 11.10.0 libpri is 1.4.14 dahdi modules are loaded with lsmod | grep dahdi lsmod | grep dahdi dahdi_transcode 6729 1 wctc4xxp dahdi_voicebus 54866 2 wctdm24xxp,wcte12xp dahdi 224134 11 dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,oct612x crc_ccitt 1717 2 wctdm24xxp,dahdi Thoughts? 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] Issue with new Te133
On Mon, Jul 14, 2014 at 12:42:34PM -0400, Jerry Geis wrote: Have a new TE133 in a dell R320. dahdi_cfg -vvv shows: DAHDI Tools Version - 2.9.1 DAHDI Version: 2.9.1 Echo Canceller(s): Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 24: D-channel (Default) (Echo Canceler: none) (Slaves: 24) 7 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Not sure why??? system.conf shows: cat system.conf loadzone=us defaultzone=us span=1,0,0,esf,b8zs bchan=1-6 dchan=24 echocanceller=mg2,1-6 I installed from source. I am running asterisk 11.10.0 libpri is 1.4.14 dahdi modules are loaded with lsmod | grep dahdi lsmod | grep dahdi dahdi_transcode 6729 1 wctc4xxp dahdi_voicebus 54866 2 wctdm24xxp,wcte12xp dahdi 224134 11 dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,oct612x crc_ccitt 1717 2 wctdm24xxp,dahdi Looking at the above, it appears you do not have the wcte13xp driver loaded. Check your /etc/dahdi/modules file and add a line to load the wcte13xp driver. 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] Issue with new Te133
Thanks, Looking at the above, it appears you do not have the wcte13xp driverloaded. Check your /etc/dahdi/modules file and add a line to load the wcte13xp driver. Correct, I had copied the OLD configs that did not have the new module. I added like you suggested and off to the races! 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] How to log caller IP address in the CDR?
can you please tell me exactly which file to edit please. -- _ -- 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] How to log caller IP address in the CDR?
Hi
Set(CDR(userfield)=${SIPPEER(${CALLERID(num),ip)})
If caller is SIP peer.
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-14 14:10 GMT-03:00 Rafael rrich...@gmail.com:
can you please tell me exactly which file to edit please.
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Re: [asterisk-users] TE131 kernel panic
On Tue, Jul 01, 2014 at 02:39:01PM -0500, Russ Meyerriecks wrote: On Tue, Jul 1, 2014 at 1:56 PM, Matt Behrens m...@zigg.com wrote: Having some issues with a TE131 apparently causing kernel panics. This is on CentOS 5.10 with DAHDI 2.9.1 (also tried 2.9.2-rc1, no difference.) There’s also an Tracebacks are very similar to this one: https://pbs.twimg.com/media/BrdrQf3CAAE3RXQ.jpg We're going to need the exact trace for the panic you're encountering. This image shows a panic involving both the te13x and an wcaxx analog card. The easiest way to do this would be to contact Digium support and they'll walk you through collecting all the info we will need to debug this. I am just closing out this thread after Matt made his system available for trouble shooting. The crash reported here is fixed in 2.9.2-rc2 in commit (a1ff3cb0 dahdi: Stop tones on channel when updating tone zone.) [1] [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=a1ff3cb0c0f3f4e65d734f4a71a090f5484505b4 The bug in dahdi was hit in this case since dahdi_cfg was called while asterisk had channels opened that were in the process of playing tones. -- 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
