Re: [asterisk-users] CallerID external call after Attended Transfer
Hi, Am Montag, den 04.02.2013, 14:45 +0100 schrieb Jonas Kellens: Hello, thanks you for your answer. The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? as Kevin already written, set this in asterisk: sendrpid=pai trustrpid=yes I don't know the T32G, but in T2x series there is a setting under Accounts-Advanced called Caller ID Header. Select PAI or PAI +FROM. The default is FROM which won't work. HTH, Karsten -- _ -- 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] Wierd question - Give me your opinion please
Minimum 3 days to Next Day Air something in. My real concern is the Analog Extensions getting hit by lightening and taking out the server, but since the cards are module based hopefully this will not happen. I may use your hot spare idea. On Mon, Feb 4, 2013 at 10:14 PM, Carlos Alvarez car...@televolve.comwrote: If you have the budget for two machines, run all services on one and keep the other for a hot backup. Rsync the configs nightly. I'm guessing that spare parts/repairs are far away from where you will be? On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley jared.bax...@gmail.comwrote: Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance. Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards. Analog extensions WILL Hit a Surge Gate before the cards, and as much precaution on grounding protection and power protection is being taken as possible. The cards WILL BE PCI not PCI-e (They are being donated) A New Dell Power-edge Server will be acquired for the PBX HERE IS MY QUESTION Would you purchase a NEW TOWER Server with PCI slots to accommodate the cards, OR Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server just for the analog extensions, I'm torn... The ease of management of one server, or the isolation of analog extensions scattered through the jungle on it's own server. Opinions? -- _ -- 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 -- 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 -- _ -- 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] Wierd question - Give me your opinion please
My gut tells me to isolate the analog from the real pbx as much as possible, but in the end it all has to be connected my the lan. I rarely use FXS devices, so i'm leary of trusting the module to burn and save the card. The closest building is 950 ft, the second is 1850 ft. These two buildings are connected via LRE's using existing 6 Pair, Unfortunately re cabling isn't an option. Other buildings are even further from the office, about 15 or so scattered about that only require 1 phone each. Doe to the flakiness of the power, the fact that they generate and have solar augmenting the service company, Homeplug and the like are not viable options. No guaranteei all the buildings are even on the same transformer I have to work within the confines of existing Telephone cabling between buildings. Perhaps one day a strong enough wireless infrastructure can be built to support these phones. Now I wonder if I can even get a NEW server with 2 PCI interfaces... On Tue, Feb 5, 2013 at 2:17 AM, Leandro Dardini ldard...@gmail.com wrote: Both the Sangoma and the Digium card are module based, so in the event of a electrical shock, the module will burn and the board will be safe. About the too long lines, the UTP cable is not the only type of cable capable of running ethernet. Skipping the too much expensive fiber optics, you can find some cheap RG58 cable with BNC connectors. 10BaseT can run up to 185 meters and you can have stations in the middle. If the distances are longer and you don't like to get your hands dirty with old equipment, then you can use the new Homeplug standard, capable of running ethernet on almost any electrical media for hundred and hundred of meters of distance. Leandro 2013/2/5 Carlos Alvarez car...@televolve.com If you have the budget for two machines, run all services on one and keep the other for a hot backup. Rsync the configs nightly. I'm guessing that spare parts/repairs are far away from where you will be? On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley jared.bax...@gmail.comwrote: Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance. Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards. Analog extensions WILL Hit a Surge Gate before the cards, and as much precaution on grounding protection and power protection is being taken as possible. The cards WILL BE PCI not PCI-e (They are being donated) A New Dell Power-edge Server will be acquired for the PBX HERE IS MY QUESTION Would you purchase a NEW TOWER Server with PCI slots to accommodate the cards, OR Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server just for the analog extensions, I'm torn... The ease of management of one server, or the isolation of analog extensions scattered through the jungle on it's own server. Opinions? -- _ -- 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 -- 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 -- _ -- 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] Wierd question - Give me your opinion please
On 5/2/13 11:45 am, Jared Baxley wrote: The closest building is 950 ft, the second is 1850 ft. These two buildings are connected via LRE's using existing 6 Pair, Unfortunately re cabling isn't an option. Other buildings are even further from the office, about 15 or so scattered about that only require 1 phone each. Have you considered running your own SHDSL between the sites, i.e. run a small DSLAM in the main building? That'd give you IP connectivity in the remote buildings which could be used for both phones and also general net access if required. You'd then avoid having to worry about analogue phones at all. A friend did this down the length of a heritage railway as they already had cable running the length of their tracks, and I believe it was fairly successful. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
2013/2/1 Don Kelly d...@donkelly.biz -Original Message- snip What I had in mind is to use someone's cellphone as a presence detector. Let me explain: - as the first thing you take along when leaving a room or location, is your own cellphone, why not use chan_mobile and a bluetooth dongle on your on PC (as you're not supposed to be within bluetooth range from an asterisk server ;-)) to advertise you're away from your desk Being completely ignorant of Bluetooth dongles, and knowing less than I should about Asterisk, I'm happy to throw in my two-cents worth. It seems that the Asterisk server need know nothing about the Bluetooth dongle--it only cares if the user is in their office. An application can run on a pc in the office that simply decides if a device it's paired with is nearby or not. Probably want to have something that makes sure it's lost a few times over a minute or so. Then the application informs the Asterisk server (a web service?) that the individual is gone. I would expect that a single pc could keep track of several people. That said, I was thinking this was a simple Bluetooth presence detection issue, so I Googled that. One result was this: http://nerdvittles.com/?p=78 This article seems updated with http://nerdvittles.com/?p=803 I read it and still questions remain: as presence detection is done though a program (inside a proximity.zip file) - on which platform can this software be installed beside CentOS ? - is it possible to install the software on a different machine from your asterisk server ? --Don -- _ -- 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] voicemail customization
Is there any solution, I want to know at first sight from which division the voicemail came from like for example I need 3 more users to send when some one calls an inbound route named darin when it reaches darin voicemail then admin should send from da...@yahoo.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
[asterisk-users] Google Voice with Asterisk 11/chan_motif
Dear Mr. Colp and/or anyone who can help, Recently Ive upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I dont have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Thanks in advance. Best regards, Josué Freitas -- _ -- 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] Wierd question - Give me your opinion please
At 08:11 PM 2/4/2013, you wrote: Would you purchase a NEW TOWER Server with PCI slots to accommodate the cards, OR Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server just for the analog extensions, Personally I think I'd purchase a number of used servers so if one dies, you have a backup or three. 25 lines does not need a modern processor and it seems silly to spend money for a warranty when you could spend less and have a number of spares. I've used these to extend ethernet over phone lines. http://www.startech.com/Networking-IO/Media-Converters/Ethernet-Extenders/10-100Mbps-VDSL2-Ethernet-LAN-Extender-Kit~110VDSLEXT Ira -- _ -- 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] Wierd question - Give me your opinion please
On Mon, Feb 4, 2013 at 11:11 PM, Jared Baxley jared.bax...@gmail.com wrote: Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance. Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards. Analog extensions WILL Hit a Surge Gate before the cards, and as much precaution on grounding protection and power protection is being taken as possible. The cards WILL BE PCI not PCI-e (They are being donated) A New Dell Power-edge Server will be acquired for the PBX HERE IS MY QUESTION Would you purchase a NEW TOWER Server with PCI slots to accommodate the cards, OR Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server just for the analog extensions, I'm torn... The ease of management of one server, or the isolation of analog extensions scattered through the jungle on it's own server. Opinions? For that number of analog stations, I would go with tried and true channel banks. Adtran or Adit would be my personal choice. I would probably consider not even using any IP phones to keep things very simple. Thanks, Steve T -- _ -- 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 and H323
Hi, Could anyone please point me to a comprehensive how-to for H323 support in Asterisk 11? I'd like to connect machines that only support H323 and Asterisk 11. I've read the h323.conf file but I'd like to see more example setups. Thanks, Vieri -- _ -- 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] Wierd question - Give me your opinion please
On Tue, Feb 5, 2013 at 11:43 AM, Ira i...@extrasensory.com wrote: Personally I think I'd purchase a number of used servers so if one dies, you have a backup or three. 25 lines does not need a modern processor and it seems silly to spend money for a warranty when you could spend less and have a number of spares. I completely agree with this. I'd buy several refurb/used machines rather than one new. We buy all of our HP refurb servers from these guys: http://www.nautilusnet.com/ alb...@nautilusnet.com -- 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] asterisk 11 and H323
Is chan_ooh323 broken in Asterisk 11? --- On Tue, 2/5/13, Vieri rentor...@yahoo.com wrote: Hi, Could anyone please point me to a comprehensive how-to for H323 support in Asterisk 11? I'd like to connect machines that only support H323 and Asterisk 11. I've read the h323.conf file but I'd like to see more example setups. Thanks, Vieri -- _ -- 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] dahdi-channels.conf parameters
Hi, I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 context = default group = 63 So what I don't get is why group is specified with 0,11. Is that groups 0 and groups 11? And then it has a random group = 63 at the end. And I dial with group 1 (dahdi/g1), but it seems to work? :) It's completely confused me as to why this actually works. hose -- _ -- 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] Google Voice with Asterisk 11/chan_motif
Josue Freitas wrote: Dear Mr. Colp and/or anyone who can help, Recently I’ve upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I don’t have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Google is responsible for sending the call to you. If you get nothing on your screen after executing xmpp set debug on and placing a call to your Google Voice number then Google is not sending the call to you. You can try restarting Asterisk to see if that makes it work. There's nothing that can be done to force them to. -- 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
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, February 05, 2013 7:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif Josue Freitas wrote: Dear Mr. Colp and/or anyone who can help, Recently I've upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I don't have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Google is responsible for sending the call to you. If you get nothing on your screen after executing xmpp set debug on and placing a call to your Google Voice number then Google is not sending the call to you. You can try restarting Asterisk to see if that makes it work. There's nothing that can be done to force them to. -- 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 -- _ -- 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] Google Voice with Asterisk 11/chan_motif
Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- 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
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Might also want to check the google hasnt detected an unusual login and is asking for the ip to be accepted. Log in to gmail with that account and check Sent from my iPhone 5 On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote: Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- 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 -- _ -- 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] dahdi-channels.conf parameters
I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 context = default group = 63 So what I don't get is why group is specified with 0,11. Is that groups 0 and groups 11? And then it has a random group = 63 at the end. And I dial with group 1 (dahdi/g1), but it seems to work? :) It's completely confused me as to why this actually works. The starting configuration file is chan_dahdi.conf which may subsequently include a dahdi-channels.conf file. The thing to remember with chan_dahdi.conf is all options are cumulative. Channels are created using the cumulative configuration when the channel = 1-23 line is processed. Anything after that line will not affect channels 1-23 since they are already created. Depending on where in the configuration the snippet you posted is found will determine if it even has any effect. 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] dahdi-channels.conf parameters
What you say...Richard Mudgett (rmudg...@digium.com): I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 context = default group = 63 So what I don't get is why group is specified with 0,11. Is that groups 0 and groups 11? And then it has a random group = 63 at the end. And I dial with group 1 (dahdi/g1), but it seems to work? :) It's completely confused me as to why this actually works. The starting configuration file is chan_dahdi.conf which may subsequently include a dahdi-channels.conf file. The thing to remember with chan_dahdi.conf is all options are cumulative. Channels are created using the cumulative configuration when the channel = 1-23 line is processed. Anything after that line will not affect channels 1-23 since they are already created. Depending on where in the configuration the snippet you posted is found will determine if it even has any effect. Richard Ah, ok. Is it possible that this file isn't even being used then? The chan_dahdi.conf is similarly terse and doesn't have an include = line: [channels] context = global_inbound switchtype = dms100 pridialplan = national signalling = pri_cpe rxgain = 1.0 txgain = 1.0 group=1 echocancel=yes channel = 1-23 jbenable=no callprogress=yes musiconhold=default usecallerid=yes hose -- _ -- 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] Google Voice with Asterisk 11/chan_motif
I indeed access Gmail and GV from a different IP than the Asterisk server, but just made it from there and it's ok. The Asterisk server is in the US but I'm currently abroad. Is that a problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert-GMAIL Sent: Tuesday, February 05, 2013 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif Might also want to check the google hasnt detected an unusual login and is asking for the ip to be accepted. Log in to gmail with that account and check Sent from my iPhone 5 On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote: Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- 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 -- _ -- 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] dahdi-channels.conf parameters
What you say...Richard Mudgett (rmudg...@digium.com): I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 context = default group = 63 So what I don't get is why group is specified with 0,11. Is that groups 0 and groups 11? And then it has a random group = 63 at the end. And I dial with group 1 (dahdi/g1), but it seems to work? :) It's completely confused me as to why this actually works. The starting configuration file is chan_dahdi.conf which may subsequently include a dahdi-channels.conf file. The thing to remember with chan_dahdi.conf is all options are cumulative. Channels are created using the cumulative configuration when the channel = 1-23 line is processed. Anything after that line will not affect channels 1-23 since they are already created. Depending on where in the configuration the snippet you posted is found will determine if it even has any effect. Richard Ah, ok. Is it possible that this file isn't even being used then? The chan_dahdi.conf is similarly terse and doesn't have an include = line: [channels] context = global_inbound switchtype = dms100 pridialplan = national signalling = pri_cpe rxgain = 1.0 txgain = 1.0 group=1 echocancel=yes channel = 1-23 jbenable=no callprogress=yes musiconhold=default usecallerid=yes If this is the contents of the chan_dahdi.conf file then yes, the dahdi-channels.conf file is not even used. Also the lines after channel = 1-23 have no effect. 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] dahdi-channels.conf parameters
What you say...Richard Mudgett (rmudg...@digium.com): What you say...Richard Mudgett (rmudg...@digium.com): I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 context = default group = 63 So what I don't get is why group is specified with 0,11. Is that groups 0 and groups 11? And then it has a random group = 63 at the end. And I dial with group 1 (dahdi/g1), but it seems to work? :) It's completely confused me as to why this actually works. The starting configuration file is chan_dahdi.conf which may subsequently include a dahdi-channels.conf file. The thing to remember with chan_dahdi.conf is all options are cumulative. Channels are created using the cumulative configuration when the channel = 1-23 line is processed. Anything after that line will not affect channels 1-23 since they are already created. Depending on where in the configuration the snippet you posted is found will determine if it even has any effect. Richard Ah, ok. Is it possible that this file isn't even being used then? The chan_dahdi.conf is similarly terse and doesn't have an include = line: [channels] context = global_inbound switchtype = dms100 pridialplan = national signalling = pri_cpe rxgain = 1.0 txgain = 1.0 group=1 echocancel=yes channel = 1-23 jbenable=no callprogress=yes musiconhold=default usecallerid=yes If this is the contents of the chan_dahdi.conf file then yes, the dahdi-channels.conf file is not even used. Also the lines after channel = 1-23 have no effect. Richard Excellent - thanks for clearing that up. -- _ -- 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 customization
On Tue, 5 Feb 2013, Darin Iv wrote: Is there any solution, I want to know at first sight from which division the voicemail came from like for example I need 3 more users to send when some one calls an inbound route named darin when it reaches darin voicemail then admin should send from da...@yahoo.com I'm not sure what you want... I'm guessing you want to send voicemail as email and set the 'from' address based on the inbound route. In voicemail.conf, you can set the program used to send email. This could be a 'wrapper' for looking up the 'from' address and constructing a command line to execute either sendmail or sendEmail.pl or whatever. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Direct dial
I have clarification in which how we can enable direct dial when we press numbers in 3cx phone on hook. Now its like we have to use dial button to dial. Previously I am able to dial directly after entering number. Now its not working. Can someone help me on it. Is this a setup that we have to do in freepbx or In 3cx Phones? Regards Darin Egocentrix -- _ -- 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
