Re: [asterisk-users] asterisk 1.8.18.1 Now Available
When will packages.asterisk.org be updated with the RPM's? Thanks EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Thursday, December 06, 2012 2:47 PM To: Asterisk Users Mailing List Subject: [asterisk-users] asterisk 1.8.18.1 Now Available The Asterisk Development Team has announced the release of Asterisk 1.8.18.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.18.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- chan_local: Fix local_pvt ref leak in local_devicestate(). (Closes issue ASTERISK-20769. Reported by rmudgett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.1 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 -- _ -- 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] updates to packages.asterisk.org?
Will there be an update to the RPM repo on packages.asterisk.org? For example http://packages.asterisk.org/centos/5/asterisk-1.8/x86_64/RPMS/ Latest is showing 1.8.15.1. Thanks EKG -- _ -- 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 1.8.15.0 Now Available
Is there an ETA on when this will show up on packages? Thanks for the work! EKG -- _ -- 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 1.8.13.0 Now Available
When will this be available at packages.asterisk.org? Thanks! EKG -- _ -- 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] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, March 05, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available On 03/05/2012 06:34 AM, Eric Germann wrote: Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories? Thanks! EKG They should be available now. -- _ -- 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] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
Well that's soon enough I guess :) Thanks for what you do! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, March 05, 2012 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes -- _ -- 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 1.8.9.2 Now Available
We update from packages. Will this make its way to packages.asterisk.org or packages.digium.com? I double checked the sites. Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Thursday, February 09, 2012 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8.9.2 Now Available The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 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: * --- Fix SIP INFO DTMF handling for non-numeric codes --- (Closes issue ASTERISK-19290. Reported by: Ira Emus) * --- Fix crash in ParkAndAnnounce --- (Closes issue ASTERISK-19311. Reported-by: tootai) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.2 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 -- _ -- 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 1.8.9.0 Now Available
We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG -- _ -- 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 1.8.9.0 Now Available
Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, January 30, 2012 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.9.0 Now Available On 01/30/2012 11:06 AM, Eric Germann wrote: We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG The RPMs should be there in a few minutes. -- _ -- 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] FreePBX not updating configs on 1.8 RPM install
Brand new instance on Centos 5.7 Installed asterisk18 via yum from RPM distribution from Digium Installed FreePBX via yum from Digium distribution. Asterisk is up. FreePBX is up. However, the changes made in FreePBX aren't written out to the config files in /etc/asterisk nor does asterisk recognize any of the configs. Am I missing something? Been a little while since I installed them, but don't recall it being this difficult. Thoughts? EKG -- _ -- 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] FreePBX not updating configs on 1.8 RPM install
Thanks. Checked. Both running as 'asterisk' EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Confirm your web server user is running as the same user as asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann Sent: Friday, December 16, 2011 3:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Brand new instance on Centos 5.7 Installed asterisk18 via yum from RPM distribution from Digium Installed FreePBX via yum from Digium distribution. Asterisk is up. FreePBX is up. However, the changes made in FreePBX aren't written out to the config files in /etc/asterisk nor does asterisk recognize any of the configs. Am I missing something? Been a little while since I installed them, but don't recall it being this difficult. Thoughts? EKG -- _ -- 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] FreePBX not updating configs on 1.8 RPM install
Answering my own question, which is probably bad form. Updated the modules to current (from 2.7.0.0), applied config, now it works. Odd. EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann Sent: Friday, December 16, 2011 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Thanks. Checked. Both running as 'asterisk' EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Confirm your web server user is running as the same user as asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann Sent: Friday, December 16, 2011 3:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FreePBX not updating configs on 1.8 RPM install Brand new instance on Centos 5.7 Installed asterisk18 via yum from RPM distribution from Digium Installed FreePBX via yum from Digium distribution. Asterisk is up. FreePBX is up. However, the changes made in FreePBX aren't written out to the config files in /etc/asterisk nor does asterisk recognize any of the configs. Am I missing something? Been a little while since I installed them, but don't recall it being this difficult. Thoughts? EKG -- _ -- 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] SIP Provider Recommendation in US
Teliax Just Works EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, March 03, 2011 11:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SIP Provider Recommendation in US -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: Thursday, March 03, 2011 10:28 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Provider Recommendation in US On 11-03-03 11:22 AM, Brent A. Torrenga wrote: I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. I've had good luck with bandwidth.com for a couple of customers running call centers. Leif. I'm happy so far with VoicePulse. Keep in mind, if you dig far enough, you'll find good and bad comments about any provider. -- _ -- 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] Voicemail on Different Server
How do you handle transfering vmail from one user to another when they're on separate servers? I'm using the single vmail server, mounted NFS partition for this right now. I'd love to be able to have them standalone so they're survivable when the WAN collapses, but I haven't figured out transfer. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Saturday, April 28, 2007 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail on Different Server Hi Steve - Can you elaborate on this, I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. This sounds like it would probably work the best, especially if you have users moving around between offices. What was so finicky and unstable about it? I am not one to quickly give up. I have found that persistence pays off when the idea is sound. Yeah, I thought I had found the silver bullet with MySQL replication (the users do float between offices, so it seemed perfect). There were a number of problems, but in the end it was table corruption as a result of the replication process that made me drop this solution. At the time I set this up, MySQL replication was really designed for one-way replication. Two way replication was possible, but required somewhat unorthodox methods. (Maybe this has changed, I don't know). Configuration is also a little tricky. It's not too bad to set it up between two machines, but 3 machines is more tricky, and 4 is even more tricky, etc, etc. This client had only 3 offices at the time, but I knew they would be expanding. They now have 6. Anyway, after getting everything working, I found that replication would periodically stop after some time. I'd have to re-create the setup, and then replication would work for a time, and then stop again later. This occurred across several different version of MySQL. I suppose I could have fixed this issue with persistence, but unfortunately this was only an annoyance compared to the major issue of data corruption. When replication worked, it was inevitable that after a time the voicemail storage table would experience data corruption. Asterisk did not handle this gracefully at all. It was effectively a total DOS. This also occurred across several versions of MySQL. Sometimes I was able to repair the tables, but usually I couldn't, and the users ended up losing quit a lot of voicemails. I did not have the ability to spend the amount of time I needed to fix the issue, so I scrapped the whole setup. Regular local voicemail storage has been flawless in all installations I've administered. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Streaming Audio Bridge
Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? Thanks Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Large number of prefixes in a route to a trunk
We're beginning to test MultiTech's CallFinder CDMA Units, one for Sprint PCS and one for Alltel. They sell both. Our intent is to use them as a backup line for our main office (which has a PRI) and a backup/911 line for our remote offices which are all connected via * over a VPN with no local trunks at any of them. In the interest of maximizing use of the lines, I'm putting together a dial plan that includes PCS-to-PCS/Nextel calling for the Sprint trunk. Essentially, the PBX would look like a cell phone to the PCS cloud. Total merged NPA-NXX list for SPCS I come up with is around 7,600 prefixes. Since our parent has offices strung out all over the US and is standardizing on SPCS, it makes sense to try and leverage as many PCS-to-PCS calls as we can. Alltel comes in at around 1940 prefixes. Has anyone found a soft limit for what * can handle in an outbound route associated with a trunk? The box that does the routing is a new quad core with 2GB of RAM. Any recommendations for whether to use the straight extensions_X.conf or write a custom dialplan with a db hook in it? I'm sort of in favor of the *.conf files since they remove an external dependency from the dialplan, if the speed is reasonable, but a prefix list like this is new territory for me. If anyone is interested, drop me an email and I will be happy to share the NPA-NXX extracts for Alltel Wireless, Sprint PCS and Nextel in CSV format. Thanks in advance and I will be happy to share with anyone or the list if there is interest in our experience with the devices (they're relatively new) EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS???
FXS cards generate ring (you connect a station to it and it rings). FXO cards sink ring (they take ring from the office). If the Octel needs ring (which it most likely does), you would need an FXS card to generate ring for it to answer. An FXO would take ring from the vmail server, which, in context, doesn't make a lot of sense (vmail doesn't call the PBX, the PBX calls vmail). EKG _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Monday, February 05, 2007 8:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS??? Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and multicore processors
I'm specing out a new box to act as a tandem switch. It will have a TE410P with 4 x PRI and support IAX connections to four other boxes using predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call routing also using gsm. No extensions actually terminate on the tandem, they're all switched to other boxes (highly distributed). On the PRI card, one goes to Embarq, the PSTN and two go to a legacy SX-200 which is being phased out. The fourth is a connection to an Adtran TSU-600 channel bank. Given this is a greenfield spec and we're building it from scratch, I'm looking at SuperMicro and their motherboards. Architecturally, I see the tandem as being CPU bound, if anything. Backbone is GigE connected to the server so I/O there isn't an issue and we aren't doing voicemail on it so it isn't diskbound. Primarily the load will in in transcoding between the PRI channels and the IAX channels. We're looking at probably no more than 100 calls simultaneously. All the remote boxes use the same codec on the channels, so it doesn't have to transcode for inter box comm's. How well does asterisk spread itself out over multiple CPU's (aka Cores). I'm looking at their 2xQuadCore (clovertown) motherboards and was spec'ing CPU's. I know this is a religious issue in some circles, but is it better to have one Quad core as fast as you can buy (4 CPU's) or 2 x Quad core at a lower speed (8 CPU's). Obviously, I've got to shoehorn a budget here and can do 8 for the price of 4 if * will spread itself out. If transcoding is threaded and doesn't deadlock for a single resource, it seems 8 cores would be better than 4. Thoughts? Thanks for any input. Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMDI support on trixbox
Last question for the day, I promise. On voip-info.org and trixbox.org, I found some old threads on MWI via SMDI. Has this been rolled into Tbox or has anyone successfully rolled it in after the fact. As part of our longterm plan, I'd like to move the legacy PBX to Tbox and pass MWI back to it via SMDI, like the current system. As we drop the extensions on the legacy side and move to all IP phones, their VM will stay with them then. Thanks EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Enterprise quality SIP provider
We LOVE Teliax. We're on a Time Warner business class fiber connection and avg 25ms latency from Ohio to Denver CO. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikas Sent: Sunday, January 28, 2007 9:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Enterprise quality SIP provider I need to setup incoming (over an 800 number and some local DID's) and outgoing phone calls (all over the country) with an Asterisk server. This asterisk server has 20 Polycom 430 phones connecting to it. I need the best possible SIP provider out there. I have tried http://www.nufone.net and http://www.broadvoice.com and they do not even come close to the expected quality. Does ATT allow companies to connect to their backbone network using SIP ? Any suggestion of companies which provide enterprise quality SIP termination and origination. The office is in a building which has a data center in the basement and has DS3 coming into the data center. I can buy as much bandwidth as I want from the data center. Regards, -- Vikas http://www.stanford.edu/~vikask/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channels Banks that support neon MWI
Anyone have suggestions for channel banks compatible with Trixbox that can set a MWI lamp on phones. We're a business, but have a lot of analog phones with the neon lamp on them and want to move them from a Mitel SX-200 to *. EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Practical limit on dial prefixes for a route
I'm aware of Cingular being GSM. We're standardizing on Sprint since Cingular is less than optimal around here. Even with LNP, knowing the NPA-NXX would nail probably 90%+ of our people. The ones that are on LNP could be added as 10 digit LCR. From a technical standpoint, can * handle over 1000+ prefixes on a route? EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Monday, January 15, 2007 9:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Practical limit on dial prefixes for a route Eric Germann wrote: Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Two comments: Cingular is GSM, Sprint is CDMA With LNP , NPA-NXX isn't enough information to determine free on network calling Since wireline to wireless LNP, the NPA assignments are no longer locked to a specific carrier. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Practical limit on dial prefixes for a route
Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Here's the kicker. Since we're on a natioinal basis, it would make sense to have a large LCR listing of prefixes reachable from the gateway, which would most likely number in the thousands of prefixes. Has anyone encountered an upper practical limit that * has for prefixes reachable via a route. I assume that search time is somewhat of a factor. The * box doing the routing is a dual core machine with 4GB of RAM, so it has lots of horsepower. Wondering what limits users have pushed it to on a large scale. Could it handle something like that or would it implode from a huge routing table (assuming our tech contacts at PCS could supply us with a national listing of NPA-NXX's on the PCS network). Thanks in advance for any info. EKG ^ depending on call volume, we may install multiple cell lines ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Practical limit on dial prefixes for a route
Correction, that's Multitech CALLFinder CDMA, not CellFinder. Sorry for the misquote. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: Monday, January 15, 2007 8:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Practical limit on dial prefixes for a route Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Here's the kicker. Since we're on a natioinal basis, it would make sense to have a large LCR listing of prefixes reachable from the gateway, which would most likely number in the thousands of prefixes. Has anyone encountered an upper practical limit that * has for prefixes reachable via a route. I assume that search time is somewhat of a factor. The * box doing the routing is a dual core machine with 4GB of RAM, so it has lots of horsepower. Wondering what limits users have pushed it to on a large scale. Could it handle something like that or would it implode from a huge routing table (assuming our tech contacts at PCS could supply us with a national listing of NPA-NXX's on the PCS network). Thanks in advance for any info. EKG ^ depending on call volume, we may install multiple cell lines ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with voicemail
I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to user, on both the VoIP and legacy system (voicemail being on a dedicated * box). 1. Thanks to jporier who can be found at ccu.edu, I figured out how to deal with MWI for all the remote servers by mounting the voicemail directory via NFS from VMAIL1 onto the VOIPx servers which host the actual phones. Then sticking a msg0.txt file into the directory makes the blinky light go on the phones. So far so good. What I'm asking the list for is either a brief code snippet or pointers to a doc/link on how to setup the following: A. None of the VOIPx servers have vmail enabled on them. When someone gets dumped to voicemail, I envision the call being transferred to the VMAIL1 server and it routing it directly to a mailbox for the user. B. VMAIL1 has no user extensions on it, just mailboxes. It gets a call on the trunk and dumps it to the appropriate vmail box based on the extension that was called. C. How do I force the vmail to go down the trunk to VMAIL1? D. How do I catch it on the other end and stick it only in a mailbox? Basically, how do I split the voicemail transfer off the local box to another? Now for a couple of architectural questions: 1. When a caller rings thru the TANDEM1 box to a VOIP1 extension, and then gets dumped to vmail, does the call go TANDEM1-VOIP1-VMAIL1 or does VOIP1 hand it off so it's only TANDEM1-VMAIL1, presuming all IAX2 trunks are running a matching subset of codecs? 2. Same thing for intracompany calls. If VOIP2 calls VOIP1 user via the tandem and gets dumped to vmail, does it go VOIP2-VOIP1-VMAIL1 or VOIP2-VMAIL1? When user is talking on PSTN over Teliax, I can see TANDEM1 doing the transcoding if necessary and bridging via IAX2 show peers. This leads me to believe it would go the former route, not the latter. If it is the former, is there a way to make it do the latter? 3. For the TANDEM1 to VMAIL1 trunk, does it make sense to do G711 as well on the trunk so it can transfer without transcoding to the voicemail box (user dials the voicemail number DID on PRI from Embarq, hits the mapping on the tandem and goes down the VMAIL1 trunk). 4. Does it make sense to have a redundant tandem running on another box and split the PRI's from the IAX trunks? Embarq is looking into forwarding the PRI DID blocks to the pilot number for the IAX2 trunk from Teliax so when it goes down or is all-trunks-busy, it comes down the 'Net pipe. Nice to have Embarq on one side of the road ariel and TW underground on the other side with separate entrances. 5. When a call is hairpinned in TANDEM1 from the Embarq PRI to the tie PRI's, is there any CPU overhead involved or is it basically done in the card, presuming matching codecs on the PRI's? Card is a digium TE405P quad PRI card. Some implementation notes: 1. All the boxes with IP addresses shown in the pic are setup. I have successful calls going Teliax - Tandem - VOIP1 and also back out to the PSTN via the Tandem. VOIP2 comes up tomorrow. PRI's are a middle of the night job later this week. 2. All are running Trixbox 2.0b2. 3. We're playing with codecs to see what gives the best quality for the bandwidth. Voip-info.org seems to point towards ilbc as having the lowest overhead, followed by gsm and g729. I presume if we want to bring fax in off the Embarq PRI and/or Teliax we're going to have to use G.711u thru to the Hylafax server with iaxmodem. Anybody have any experience with bringing fax in over a IAX2 trunk from Teliax (or any other voip provider for that matter)? We're switching this Thursday to a 10Mbps symmetric fiber connection from Time Warner Business Class. Once I get this working, I'm willing to write up a how-to (I'm going to have to anyways for documentation, just needs to be sanitized) and put a pointer or the doc on voip-info.org Thanks in advance. EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
