Re: [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?
Hello Armin Schindler, Am 2012-06-28 09:52:30, hacktest Du folgendes herunter: On 27.06.2012 18:46, Michelle Konzack wrote: Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus? Which PCI-ID is that? I do not know, because I have not bougth it yet. It is only named: Eicon DIVA Server 4BRI-8M 2.0 800-665-02 version 2.0 Waiting for a BeroNet (HFC Chipset) too... Armin Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- 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] BT Fibre and 2701HGV
Hi Does anyone have any experience of connecting SIP phones to an asterisk server through the 2701HGV router that BT supply with their Infinity product? Thanks in Advance Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Please dont tell me this is impossible
I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI-command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing until $maxdigits is received or $timeout has expired. So far I found impossible to achieve this functionality. Am I missing something? Philip -- _ -- 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] Please dont tell me this is impossible
Am 29.06.2012 11:38, schrieb CDR: I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI-command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing until $maxdigits is received or $timeout has expired. So far I found impossible to achieve this functionality. Am I missing something? Philip The Playcommand will be interrupted by the key but the agi result contains the offset. So you can play this file from offset again until you $maxdigits has been pressed. Take a look here: https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE -- _ -- 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] PRI trunk between Asterisk servers does not work.
In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Dahdi Dropping Calls
Hi Guys Has anyone seen Dahdi dropping incoming calls with Hangup cause 27? It only drops whilst we are on the phone? Its not every single call Any ideas? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield Sent: 29 June 2012 01:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI trunk between Asterisk servers does not work. In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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:
Re: [asterisk-users] Dahdi Dropping Calls
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27? Please don't hijack a thread. Start a new message with your question, since it'll screw up message threading. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Please dont tell me this is impossible
This is from the documentation of Perl-AGI
$AGI-stream_file($filename, $digits, $offset)
Executes AGI Command STREAM FILE $filename $digits [$offset]
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI-stream_file('demo-echotest', '0123');
$AGI-stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed
It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip
On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip
The Playcommand will be interrupted by the key but the agi result contains
the offset. So you can play this file from offset again until you $maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE
--
_
-- 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] Please dont tell me this is impossible
On 12-06-29 05:38 AM, CDR wrote: I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI-command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing until $maxdigits is received or $timeout has expired. So far I found impossible to achieve this functionality. Am I missing something? Philip Just use an existing library, rather then rolling your own. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] PRI trunk between Asterisk servers does not work.
Quoting Ioan Indreias indre...@gmail.com: On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 Hello Ernie, Could you post the dahdi/system.conf from both voip1 and voip3 servers? I suspect that you have not correctly defined the data channel (dchan setup should be in system.conf and not in chan_dahdi.conf, where I see a not necessarily dchannel configuration) HTH, Ioan Okay, here's /etc/dahdi/system.conf (it's unmodified from the autogenerated file): # Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 26 22:53:04 2010 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,0,esf,b8zs # termtype: te bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 span=3,0,0,esf,b8zs # termtype: te bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 span=4,0,0,esf,b8zs # termtype: te bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global data loadzone = us defaultzone = us This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join
Re: [asterisk-users] Dahdi Dropping Calls
I've never seen this on incoming calls, only outgoing calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Colin Sent: Friday, June 29, 2012 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dahdi Dropping Calls Hi Guys Has anyone seen Dahdi dropping incoming calls with Hangup cause 27? It only drops whilst we are on the phone? Its not every single call Any ideas? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield Sent: 29 June 2012 01:52 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI trunk between Asterisk servers does not work. In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? It looks like this: # asterisk -rx 'pri show span 3' Primary D-channel: 72 Status: Provisioned, Down, Active Switchtype: National ISDN Type: Network Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 Logical Channel Mapping: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 The only differences I see between 'pri show span 3' and 'pri show span 4' are that the status on span 4 is Provisioned, Up, Active and that the D-channel is different, which is to be expected. It might be useful to see the complete chan_dahdi.conf from Voip1. To save space, you can list it without comments like this: # grep -v '^;' /etc/asterisk/chan_dahdi.conf Okay, here you go: [channels] usecallerid=yes cidsignalling=bell cidstart=polarity facilityenable=yes hidecallerid=no callwaitingcallerid=yes callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=no immediate=no group=1 signalling=pri_cpe switchtype=national pridialplan=unknown relaxdtmf=yes context=local channel=1-23 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=3600 #include dahdi-channels.conf And dahdi-channels.conf looks like: group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group=4 context=default switchtype = national signalling = pri_net channel = 73-95
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
Quoting Tony Mountifield t...@softins.co.uk: In article 4feccd0c.1020...@fivecats.org, James Sharp ja...@fivecats.org wrote: On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). No, the group=63 lines are actually redundant. It is the settings *above* each channel= line that get applied to the channels when they are created. To the OP: what does pri show span 3 give you on Voip1? Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] PRI trunk between Asterisk servers does not work.
- Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) --Tim -- _ -- 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] IAX Trunk issue. (Dale Noll
Dale,
Sorry for taking so long to answer, I've been traveling.
Thanks so much for the suggestion, your solution worked perfectly. I'm not
sure why I didn't notice that the IAX trunk was working in the other direction.
Once again, thanks for your help.
Mitch
Date: Mon, 25 Jun 2012 05:44:37 -0500
From: Dale Noll dn...@wi.rr.com
Subject: Re: [asterisk-users] IAX Trunk issue.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: 4fe84115.60...@wi.rr.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 06/24/2012 07:53 PM, Mitchell Johnson wrote:
I'm testing a few IAX trunk scenarios in a controlled lab. From server2
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes
across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of
ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial
6099 it also plays tt-weasels as it's supposed to, but it's not the
tt-weasels under its extension. It also dials the s extension.
I only placed the s extension in the dial plan to verify that the traffic was
going across the IAX trunk and hitting the correct context.
Any help would be greatly appreciated.
Thanks Mitch
[phones]
exten = _60XX,1,Dial(IAX2/trunk-1)
exten = _X.,1,Dial(IAX2/trunk-1)
exten = 5000,1,Dial(SIP/${EXTEN})
exten = 5000,n,Hangup
same = n,Hangup()
exten = 5099,1,Playback(tt-monkeys)
exten = 5099,n,HangUp
You are not telling asterisk-1 where you want the call to go, so it is going to
's'.
Try adding the extension to the Dial() command on asterisk-2. Change
Dial(IAX2/trunk-1)
to
Dial(IAX2/trunk-1/${EXTEN})
Note: It appears that you are doing it correctly from asterisk-1
towards asterisk-2
exten = _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
Assuming, of course, that the variable IAXTrunk is properly set.
Dale
--
The truth speaks for itself. I'm just the messenger.
Lyta Alexander - Babylon 5
--
_
-- 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] PRI trunk between Asterisk servers does not work.
On 12-06-29 11:40 AM, Tim Nelson wrote: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) *CLI pri show version -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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] Please dont tell me this is impossible
Please use more meaningful subjects. 'Please dont tell me this is impossible' I'm sure there are lots of things I could tell you that are impossible. 'Issue with Perl-AGI' I'm sure there are many issues with Perl and AGI if you don't understand the protocol. Better subjects = better responses. -- 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
Re: [asterisk-users] BT Fibre and 2701HGV
On 29/6/12 9:59 am, Ishfaq Malik wrote: Does anyone have any experience of connecting SIP phones to an asterisk server through the 2701HGV router that BT supply with their Infinity product? Good luck with that. The BT 'Home Hub' and 'Business Hub' routers they supply with retail ADSL and FTTC products seem to have a very poorly written SIP ALG in them that cannot be disabled [0] easily. This seems to play havoc with any attempt to hook up SIP devices - even just one handset. [0] I have read that it's possible to disable the ALG through a telnet session to the router, but it's somewhat fiddly and doesn't always 'stick' - so has to be repeated whenever the router is restarted. In my experience it's far easier just to replace the router with something competent. 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] PRI trunk between Asterisk servers does not work.
Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 Dropping Calls
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27? It only drops whilst we are on the phone? Its not every single call Any ideas? Libpri can generate that cause code when T309 expires. T309 starts when the link goes down. When T309 expires, active calls are dropped because the link did not return. 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] PRI trunk between Asterisk servers does not work.
- Original Message - Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. Yep, that'd cause what you're seeing. Glad we could help. :) --Tim -- _ -- 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] VoIP Company looking for Asterisk/VoIP Engineer
Hi, I work for a VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk/VoIP to help work on the following: - Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin tasks - Devise and implement scalability strategies so that adding additional capacity is easy and does not compromise anything about the current system - Troubleshooting call quality issuses through our network (jitter, audio dropouts..) Candidates should have the following experience: - Minimum 3 years working with VoIP/Asterisk - Have worked in an environment with a significant number of phones (500) - Experience working with Cisco networking devices - QoS knowledge is a huge plus. Having experience with VoIP over carrier-class wireless links is a definite plus. This is a part-time contractor position. We are located in Southern California, and while having someone local would be ideal, telecommuting is an option. Hourly rate DOE. Please email all resumes directly to me at jlama...@gmail.com Thank you. -- _ -- 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] .lock file issue
voice mail folder, I saw a .lock file. Apparently this was caused by a core dump in the mail module. I witnessed this just a bit ago. There are core files in /tmp. I'll search Jira for outstanding tickets this weekend and open one if not found. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] .lock file issue
Doug: You may want to apply the patch on ASTERISK-19923 - it fixes a critical problem in app_voicemail in the latest version. We are planning on releasing a new version of 1.8.13/10.5, which will include this patch. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org - Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 29, 2012 3:05:02 PM Subject: Re: [asterisk-users] .lock file issue voice mail folder, I saw a .lock file. Apparently this was caused by a core dump in the mail module. I witnessed this just a bit ago. There are core files in /tmp. I'll search Jira for outstanding tickets this weekend and open one if not found. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Intro to DECT vs IP
We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable through SIP) etc? Thanks -- _ -- 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] .lock file issue
You may want to apply the patch on ASTERISK-19923 - it fixes a critical Thank you for the info, I'll apply it this weekend! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Intro to DECT vs IP
On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis mdup...@ocg.ca wrote: We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? The SIP side of every DECT phone I've worked with looks/works just like any regular SIP phone. Asterisk just sees a SIP endpoint. If it's multi-handset/multi-account it's much like configuring a multi-line SIP phone. Can you push configuration info to individual phones? (Are they individually addressible / configurable through SIP) etc? This is all dependent on the phone/base, but every one I've used does. Again, it works just like any other SIP handset that supports a central config server. Honestly while there's a little bit of learning to do, deploying a SIP-DECT solution isn't really different from other phones and you should just jump into it. We are very pleased with Spectralink for larger/industrial applications and Panasonic for small office applications. Devote four hours to learning and you'll be comfortable with the configs for either. -- 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] Intro to DECT vs IP
Can you really mix match any base station with any DECT handset? Do handsets have proprietary features which only work with their own basestations? (eg: transfer between handsets)? Can i buy a good base station and get cheap Costco Dect handsets? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez [car...@televolve.com] Sent: Friday, June 29, 2012 4:58 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? The SIP side of every DECT phone I've worked with looks/works just like any regular SIP phone. Asterisk just sees a SIP endpoint. If it's multi-handset/multi-account it's much like configuring a multi-line SIP phone. Can you push configuration info to individual phones? (Are they individually addressible / configurable through SIP) etc? This is all dependent on the phone/base, but every one I've used does. Again, it works just like any other SIP handset that supports a central config server. Honestly while there's a little bit of learning to do, deploying a SIP-DECT solution isn't really different from other phones and you should just jump into it. We are very pleased with Spectralink for larger/industrial applications and Panasonic for small office applications. Devote four hours to learning and you'll be comfortable with the configs for either. -- 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] Intro to DECT vs IP
On 29/6/12 11:16 pm, Michelle Dupuis wrote: Can you really mix match any base station with any DECT handset? Yes and no. Do handsets have proprietary features which only work with their own basestations? (eg: transfer between handsets)? Yes. And that's the 'no' part of my answer above - whilst they may make take calls, you might well lose additional functionality. Transfer hasn't been a particular problem (in my experience, it's better to use the native asterisk functions for this on DECT phones), but call lists most definitely have been an issue. Can i buy a good base station and get cheap Costco Dect handsets? As above, if you weren't worried about all the features, quite probably. But reasonable Gigaset DECT handsets designed for the base aren't exactly expensive - I think the C610H is around the 30GBP mark - substantially less if you're ordering quantity. And I've seen older models for substantially less - I picked up a batch of new - but old model S450s for around 30GBP for 6. I don't think I've seen DECT units in Costco for much less than 20 GBP. 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] Intro to DECT vs IP
I like the look of the C610H. Is there a matching DECT base station by Gigaset? (I can't figure this out looking at their site) I see a C610IP but it's not clear if that base station supports multiple SIP accounts, multiple calls active. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall [aster...@lists.minotaur.cc] Sent: Friday, June 29, 2012 6:27 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On 29/6/12 11:16 pm, Michelle Dupuis wrote: Can you really mix match any base station with any DECT handset? Yes and no. Do handsets have proprietary features which only work with their own basestations? (eg: transfer between handsets)? Yes. And that's the 'no' part of my answer above - whilst they may make take calls, you might well lose additional functionality. Transfer hasn't been a particular problem (in my experience, it's better to use the native asterisk functions for this on DECT phones), but call lists most definitely have been an issue. Can i buy a good base station and get cheap Costco Dect handsets? As above, if you weren't worried about all the features, quite probably. But reasonable Gigaset DECT handsets designed for the base aren't exactly expensive - I think the C610H is around the 30GBP mark - substantially less if you're ordering quantity. And I've seen older models for substantially less - I picked up a batch of new - but old model S450s for around 30GBP for 6. I don't think I've seen DECT units in Costco for much less than 20 GBP. 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 -- _ -- 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] Intro to DECT vs IP
On 30/6/12 12:12 am, Michelle Dupuis wrote: I like the look of the C610H. Is there a matching DECT base station by Gigaset? I use the N300IP. Supports 3 active SIP calls I believe - and yes, does have multiple SIP accounts (6, if I recall correctly). 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] Intro to DECT vs IP
Do the C610H and C300IP use an international standard for frequencies? I can't even find gigaset sold in USA/Canada... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall [aster...@lists.minotaur.cc] Sent: Friday, June 29, 2012 8:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On 30/6/12 12:12 am, Michelle Dupuis wrote: I like the look of the C610H. Is there a matching DECT base station by Gigaset? I use the N300IP. Supports 3 active SIP calls I believe - and yes, does have multiple SIP accounts (6, if I recall correctly). 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 -- _ -- 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
