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Hi To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk system, we are looking to buy an ISDN call generator/simulator device. The minimum requirements include: - Not too expensive - PRI support (BRI support is a plus) - CCS+CRC4 farming + HDB3 coding - EuroISDN (DSS1) support. - A minimum of 4 ports (120 channels/concurrent calls) - Compatibility with Digium cards. - DUT in TE

Sorry, apparently I forgot that we are looking at the Monday before Devcon for this trip. > ------------------------------ > > Message: 4 > Date: Wed, 24 Jun 2015 11:59:29 +0300 > From: Eric Klein <eric.klein at greenfieldtech.net> > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Organizing a Pre Astricon road trip > Message-ID: > <CAD4dZ0= > KfD5Nf8t5cFEuuHr10ip+YKJA_cMM2TgfZgWpCwe3kg at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > >...

...will vary, we will do our very best to accommodate your guests General admissions required So please let me know if you are interested, as we get closer we will discuss total price and how people can pay it. Looking forward to seeing you in Orlando in October. Eric -- Eric Klein Sr. Consultant GreenfieldTech Mobile +972-54-666-0933 Email Eric.Klein at greenfieldtech.net -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150624/12eac2a4/attachment.html>

...arted> > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > -- > Eric Klein > Sr. Consultant > GreenfieldTech > Mobile +972-54-666-0933 > Email Eric.Klein at greenfieldtech.net > Skype: EricLKlein > Web: http://www.greenfieldtech.net/ > > *Disclaimer:* > This e-mail is intended solely for the person to whom it is addressed and may contain confidential or legally privileged information....

Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the carrier) with the ability to go to pass through should the Asterisk PBX (software or hardware level) fail. I did not see this feature in the Digium, Sangoma, Allo, or

...------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://list...

...------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists...

...user=phone option (Reported by Matt Jordan) * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for connection-oriented transports. (Reported by Matt Jordan) * ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-24678 - [PATCH] Added atxfer* settings to features.conf.sample (Reported by Niklas Larsson) * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported by cloos) * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by Dan Jenkins)...

...user=phone option (Reported by Matt Jordan) * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for connection-oriented transports. (Reported by Matt Jordan) * ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-24678 - [PATCH] Added atxfer* settings to features.conf.sample (Reported by Niklas Larsson) * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported by cloos) * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by Dan Jenkins)...

...ed by John Fawcett) * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-26864 - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) * ASTERISK-26846 - chan_sip: Add rtcp-mux support (Reported by Sean Bright) * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions...

...ed by John Fawcett) * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-26932 - [patch] SIP/SDP: No rtpmap for static RTP payload IDs (Reported by Alexander Traud) * ASTERISK-26864 - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) * ASTERISK-26846 - chan_sip: Add rtcp-mux support...

...ed by John Fawcett) * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-26932 - [patch] SIP/SDP: No rtpmap for static RTP payload IDs (Reported by Alexander Traud) * ASTERISK-26864 - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) * ASTERISK-26846 - chan_sip: Add rtcp-mux support...

...4575 - [patch]Make capath work for res_pjsip (Reported by cloos) * ASTERISK-24678 - [PATCH] Added atxfer* settings to features.conf.sample (Reported by Niklas Larsson) * ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by Matt Jordan) * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for connection-oriented transports. (Reported by Matt Jordan) * ASTERISK-24553 - ARI/AMI: Include language in standard channel...

...4575 - [patch]Make capath work for res_pjsip (Reported by cloos) * ASTERISK-24678 - [PATCH] Added atxfer* settings to features.conf.sample (Reported by Niklas Larsson) * ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-24351 - [patch] Allow passing options and command to MixMonitor when recording in ConfBridge (Reported by Gareth Palmer) * ASTERISK-24553 - ARI/AMI: Include language in standard channel snapshot output (Reported by Matt Jordan) * ASTERISK-24552 - ARI...

...vestigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) - [ASTERISK-26427 <https://issues.asterisk.org/jira/browse/ASTERISK-26427>] - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) - [ASTERISK-26932 <https://issues.asterisk.org/jira/browse/ASTERISK-26932>] - [patch] SIP/SDP: No rtpmap for static RTP payload IDs (Reported by Alexander Traud) - [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>] - res_pjsip_session: A...