Re: [asterisk-users] How to turn on the H323 logging on Asterisk
Sema Arca wrote: Hi Richard, I could not succeed to make my ooh323 work somehow. I can see the peers and the users but although my exten definition states that the call should be forwarded to a GK, Asterisk does not send it out. I also have the same problem with registration. Do you think you can give me some ideas? Maybe send your conf as a reference? I am sorry I have no experience using ooh323 with a gatekeeper. My setup is as an endpoint between asterisk and a Panasonic TDM100 PBX. Regards, Richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to turn on the H323 logging on Asterisk
Can you still send the config files? Maybe I can come up with an idea? :( On Mon, Jun 16, 2008 at 10:32 AM, Richard Scobie [EMAIL PROTECTED] wrote: Sema Arca wrote: Hi Richard, I could not succeed to make my ooh323 work somehow. I can see the peers and the users but although my exten definition states that the call should be forwarded to a GK, Asterisk does not send it out. I also have the same problem with registration. Do you think you can give me some ideas? Maybe send your conf as a reference? I am sorry I have no experience using ooh323 with a gatekeeper. My setup is as an endpoint between asterisk and a Panasonic TDM100 PBX. Regards, Richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to turn on the H323 logging on Asterisk
Sema Arca wrote: Can you still send the config files? Maybe I can come up with an idea? :( extensions.conf entry exten = _1XX,1,Dial(OOH323/[EMAIL PROTECTED]) exten = _1XX,2,Congestion ooh323.conf [general] h323id=ObjSysAsterisk e164=100 callerid=asterisk context=default tos=lowdelay disallow=all allow=alaw dtmfmode=inband [Panasonic] type=friend context=default ip=192.168.0.2 port=1720 disallow=all allow=alaw Regards, Richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents getting stuck busy
Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be stuck as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running Asterisk 1.4.17, and the bug sounds a lot like http://bugs.digium.com/view.php?id=9618 but that bug looks to be fixed in 1.4.17. -- Kyle Sexton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfers with TE12xp
Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks for the link. I think I will be using this product. Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: Saturday, June 14, 2008 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. 2008/6/12 Syed Nasruddin [EMAIL PROTECTED]: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Our clients use this for E1 Pri: http://www.voicetronix.com/logger.htm Not sure if there is a analogue solution. -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers with TE12xp
More info about the problem. This occurs, when I try to transfer using the *2 funcionality into aterisk Thanks 2008/6/16 voip crazy [EMAIL PROTECTED]: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help! - Double NAT issue
Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically works fine. - The wife and I are having lots of fun playing around with all the VOIP phones I'm using to talk to the thing. Now. - I want to try to take a phone to my office and try to connect from there. - But I can't. - Sound familiar? Here's my setup some scenarios: |---Home-||-OFFICE---| Asterisk box Linksys WRT54G---Internet---Linksys WRT54G At home... Router: Linksys WRT54G Public IP address: 61.25.172.48 (static) Public Netmask: 255.255.255.128 DNS1: 220.152.38.233 DNS2: 220.152.38.201 Internal IP range: 10.0.0.xxx Internal IP Netmask:255.255.255.0 Router's internal IP: 10.0.0.1 DMZ Enabled, points to: 10.0.0.12 (Asterisk server) (see below) DHCP Enabled, pool starts: 10.0.0.100 Asterisk Server IP address: 10.0.0.12 Netmask:255.255.255.0 DNS:Same as router's. Changes made to sip.conf externip=61.25.172.48 localnet=10.0.0.0/255.255.255.0 nat=yes FYI - No other changes made to ANY of Asterisk's .conf files. - It's a basic 'vanilla' test box. At the office... Router:Linksys WRT54G (out of the box config) Scenarios: I have a Sipura SPA-1001, Cisco-7940, Cisco-7905, and X-Lite running on my home PC. Although I've got DNS servers assigned, I'm not using server.domain names (IP addresses only). - So I believe DNS is not an issue. Scenario A. - When the devices are 'pointing' to the Asterisk server's 'internal' IP (10.0.0.12), they all register and work fine. Scenario B. - If I configurer a phone to use (as a proxy) the home's 'public IP' (61.25.172.48), it works fine. - This tells me (I believe) that the phone is going to the router's 'public IP' but since DMZ is turned on, all the ports are forwarded to the Asterisk box's 'internal' IP (10.0.0.12). Scenario C. - The problem... If I take a device to my office (i.e. the Sipura) and connect it. - It is configured to 'talk' to my home's 'public IP'. - This thing doesn't even REGISTER with the Asterisk server. - So I can't even try to make a call. This is verified (from the office) by being telnet(ted) into my home Asterisk box and watching it's console. Anybody have any clue? If you want to try for yourself, set up a device and try to connect to my box's 'public IP' (above) and use a username of '60' with a password of '1234'. - If that works, try extension '1000' and see if you get the Asterisk box's 'congratulations' message. I'd be very interested in your results. Also, if anybody wants to take this off-forum and discuss/help me out, I'll be greatly thankful. - I have a Broadvoice account and we can even establish a phonecon. Thanks VERY MUCH in advance. Gary Guthary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! - Double NAT issue
On Mon, Jun 16, 2008 at 7:33 AM, Gary Guthary [EMAIL PROTECTED] wrote: Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically works fine. - The wife and I are having lots of fun playing around with all the VOIP phones I'm using to talk to the thing. Now. - I want to try to take a phone to my office and try to connect from there. - But I can't. - Sound familiar? Here's my setup some scenarios: |---Home-||-OFFICE---| Asterisk box Linksys WRT54G---Internet---Linksys WRT54G At home... Router: Linksys WRT54G Public IP address: 61.25.172.48 (static) Public Netmask: 255.255.255.128 DNS1: 220.152.38.233 DNS2: 220.152.38.201 Internal IP range: 10.0.0.xxx Internal IP Netmask:255.255.255.0 Router's internal IP: 10.0.0.1 DMZ Enabled, points to: 10.0.0.12 (Asterisk server) (see below) DHCP Enabled, pool starts: 10.0.0.100 Asterisk Server IP address: 10.0.0.12 Netmask:255.255.255.0 DNS:Same as router's. Changes made to sip.conf externip=61.25.172.48 localnet=10.0.0.0/255.255.255.0 nat=yes FYI - No other changes made to ANY of Asterisk's .conf files. - It's a basic 'vanilla' test box. At the office... Router:Linksys WRT54G (out of the box config) Scenarios: I have a Sipura SPA-1001, Cisco-7940, Cisco-7905, and X-Lite running on my home PC. Although I've got DNS servers assigned, I'm not using server.domain names (IP addresses only). - So I believe DNS is not an issue. Scenario A. - When the devices are 'pointing' to the Asterisk server's 'internal' IP (10.0.0.12), they all register and work fine. Scenario B. - If I configurer a phone to use (as a proxy) the home's 'public IP' (61.25.172.48), it works fine. - This tells me (I believe) that the phone is going to the router's 'public IP' but since DMZ is turned on, all the ports are forwarded to the Asterisk box's 'internal' IP (10.0.0.12). Scenario C. - The problem... If I take a device to my office (i.e. the Sipura) and connect it. - It is configured to 'talk' to my home's 'public IP'. - This thing doesn't even REGISTER with the Asterisk server. - So I can't even try to make a call. This is verified (from the office) by being telnet(ted) into my home Asterisk box and watching it's console. Anybody have any clue? If you want to try for yourself, set up a device and try to connect to my box's 'public IP' (above) and use a username of '60' with a password of '1234'. - If that works, try extension '1000' and see if you get the Asterisk box's 'congratulations' message. I'd be very interested in your results. Also, if anybody wants to take this off-forum and discuss/help me out, I'll be greatly thankful. - I have a Broadvoice account and we can even establish a phonecon. Thanks VERY MUCH in advance. Gary Guthary I just tried and it timed out. Is SSH or HTTP running on your box? I cannot access those either. I get a telnet login. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! - Double NAT issue
On Mon, 16 Jun 2008, Gary Guthary wrote: If I take a device to my office (i.e. the Sipura) and connect it. - It is configured to 'talk' to my home's 'public IP'. - This thing doesn't even REGISTER with the Asterisk server. - So I can't even try to make a call. This is verified (from the office) by being telnet(ted) into my home Asterisk box and watching it's console. Anybody have any clue? If you want to try for yourself, set up a device and try to connect to my box's 'public IP' (above) and use a username of '60' with a password of '1234'. - If that works, try extension '1000' and see if you get the Asterisk box's 'congratulations' message. I'd be very interested in your results. Works just fine. Tell the remote phone to use a STUN server. (And you might want to set dtmf=rfc2388 mode in sip.conf) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers with TE12xp
On Mon, Jun 16, 2008 at 6:39 AM, voip crazy [EMAIL PROTECTED] wrote: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've pasted the zapata config for the PRI line, please tell me what could be wrong and the cause my calls hangs up. Any clue will be welcomend. Best Regards. VoipCrazy -- /etc/asterisk/zapata.conf --- language=es context=from-zaptel relaxdtmf=yes signalling=pri_cpe signallingtype=euroisnd rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 ;callgroup=1 ;pickupgroup=1 immediate=no ;busydect=yes busycount=6 faxdetect=both group=0 channel=1-15,17-31 I don't see anything obviously wrong with the above. How about some verbose output from the Asterisk CLI? If that doesn't shed some light on it, how about pri debug span 1 output? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astribank and Celular Interface Module
Hi, I have a Xorcom Astribank connected to my Asterisk server. In one of the Astribanks FXO port I have a Celular Interface Module. My problem is the Astribank is receiving a early answer from the module, which doesn't happen with a ATA connected to the same module. This is causing some trouble with my billing system. I already tried the answeronpolarityswitch option, what else can I do ? Thanks for your help, -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! - Double NAT issue
Gordon Henderson wrote: On Mon, 16 Jun 2008, Gary Guthary wrote: If I take a device to my office (i.e. the Sipura) and connect it. - It is configured to 'talk' to my home's 'public IP'. - This thing doesn't even REGISTER with the Asterisk server. - So I can't even try to make a call. This is verified (from the office) by being telnet(ted) into my home Asterisk box and watching it's console. Anybody have any clue? If you want to try for yourself, set up a device and try to connect to my box's 'public IP' (above) and use a username of '60' with a password of '1234'. - If that works, try extension '1000' and see if you get the Asterisk box's 'congratulations' message. I'd be very interested in your results. Works just fine. Tell the remote phone to use a STUN server. (And you might want to set dtmf=rfc2388 mode in sip.conf) Gordon Works fine from here too. Bails -- This message has been scanned for viruses and dangerous content by MailScanner at Circlemail, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! - Double NAT issue
Just FWIW, I have been doing double NAT with asterisk and all kinds of SIP phones for years, including BT101, Sipura 941, and Polycom IP500 plus many cheap no name brands, plus many softphones like Zoiper, X-Lite and Gizmo Project.. However, DMZ has never worked properly for me with asterisk on any router. No idea why, but I forward ports to asterisk rather than use a DMZ. YMMV ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres
Hi, I'm having trouble with a TE220p PRI card and (outbond) caller identification. Previously with usecallingpres=no everything was Ok, one small difference between the BRI (B410p) was that the callerid needed to be stripped from one number. But then came the need to make hidden calls, and so to enable usecallingpres and SetCallerPres(). if running SetCallerPres(prohib) then the end call get 'private number' which is what we want but with SetCallerPres(allowed) or SetCallerPres(allowed_not_screened) i'm not able to get the number i want, i got the line identification number in any case (even with a callerid of 10 digits) any idea ? my zapata.conf: 8 [trunkgroups] [channels] context=from-rnis-t2-dys language=fr switchtype=euroisdn signalling=pri_cpe callwaiting=no threewaycalling=no callprogress=no busydetect=no pridialplan=unknown prilocaldialplan=dynamic priindication=outofband internationalprefix=00 nationalprefix=33 localprefix= privateprefix= unknownprefix= relaxdtmf=yes hidecallerid=no usecallingpres=yes echocancel=yes faxdetect=incoming immediate=no group=1 channel = 1-15 8 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote: Is there a contradiction between them? Naw; Steve's just showin' his ass again. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On June 15, 2008 12:04:01 pm randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! Before you ask, it's 1.0 I think. ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote: Is there a contradiction between them? Naw; Steve's just showin' his ass again. Cheers, -- jra Nah, just showing various tactics, sure some contradict each other. It depends on what level you attain Please read up, I will be glad to educate you more when you are ready Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents getting stuck busy
On Mon, Jun 16, 2008 at 12:30 PM, Kyle Sexton [EMAIL PROTECTED] wrote: Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be stuck as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running Asterisk 1.4.17, and the bug sounds a lot like http://bugs.digium.com/view.php?id=9618 but that bug looks to be fixed in 1.4.17. I could suggest you trying on latest version (currently 1.14.21) or at least try this patch http://bugs.digium.com/view.php?id=12127 The description doesn't match your issue, however there was found old code handling dialstatus and translating it to agent state, which could be cause of your problem. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
He is talking about a free Digium Screwdriver On Mon, Jun 16, 2008 at 11:07 AM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 15, 2008 12:04:01 pm randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! Before you ask, it's 1.0 I think. ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Brian, I agree with you. Google has all the answers, but not the experience. The reason I use lists is to get opinions 'experienced' users. From experience, product manuals say one thing and when the rubber meets the road, its a different story. Thanks though for your kind comments Brian J. Murrell wrote: On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote: On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote: Please advice on channel bank Dude. There's the cool new website you should check out. It's www.google.com. Seriously. This list is not full of people waiting to do the simplest research at your request. Spend a few minutes and do some self-help before coming here asking the simplest, most general questions. You are more likely to get answers to interesting questions rather than mundane-google-would-have-told-you-all-you-need-to-know-in-5-minutes questions. b/ While true to some degree, I assumed he was looking for someone to recommend a certain product based on good experiences in the Asterisk World. See, I saw the quotes around channel bank more as the follow question what is a channel bank. Maybe it's a language thing and perhaps the OP can take as constructive criticism to be more to one's actual point when asking a question. If he really did understand what a channel bank is and was looking for recommendations, something more direct like Any recommendations on which channel bank(s) I should consider using? would have been much more fruitful I suspect. Google may be good for getting information but will turn up a good many ads too. Most of these ads/sites all claim to be the best. We are the leaders of (such and such) Sure, but all of them will give him a good idea of what one actually is, which is really what I suspect the question was. An obvious pitfall I met was Citel gateways. Maybe they have improved for the Definity line, but going that route a year and a half ago made me look very bad. I wish I had asked on the list and got someone with some experience to say, think twice. Agreed. I wholeheartedly agree with soliciting for and giving product recommendations and experiences, but questions like what is ... most likely can almost always be answered from google with a little effort on one's own behalf. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Request to mailing list asterisk-users rejected
thanks moderatorit was a perfectly reasonable email - just oversized because of all the urls. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Mon 6/16/2008 1:55 PM To: Dean Collins Subject: Request to mailing list asterisk-users rejected Your request to the asterisk-users mailing list Posting of your message titled you gotta think like batman and have many tools... has been rejected by the list moderator. The moderator gave the following reason for rejecting your request: No reason given Any questions or comments should be directed to the list administrator at: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Request to mailing list asterisk-users rejected
On Mon, 2008-06-16 at 15:01 -0400, Dean Collins wrote: thanks moderatorit was a perfectly reasonable email - just oversized because of all the urls. No reason given This is most likely my fault... Just for clarification, any message over 40k gets moderated. I thought I told the mailing list system to actually *explain* why I rejected the message, but it appears I might have been a little too click-happy when I did it. Sorry for any confusion this might have caused... -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Sun, 2008-06-15 at 18:04 +0200, randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! By tweaker, I assume you mean the small screwdrivers we often give away. I don't go anywhere without one (and I've even found that airport security tends not to take them away from me if I separate the plastic handle from the metal piece). I'm glad we were able to save your bacon. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
Hi Steve and the rest of the list, On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T My Asterisk Server has two NIC with a channel bonding setup (Balance TLB) connected to the same switch, and it does not have any firewall rule. I'm attaching a file with the output of sip set debug on the CLI of a call in this situation. Although calls made with SIP phones have this strange behavior, when I place a call with an analog phone connected to a FXS port of the same TDM card (see below for full description) this does not happen. Thanks, any help will be really appreciated... -- Nacho Linux Counter #156439 On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users SIP-Debug-143.txt.gz Description: GNU Zip compressed data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Euro_isdn PRI Line, callerid and usecallingpres
Benoit Plessis a écrit : Hi, I'm having trouble with a TE220p PRI card and (outbond) caller identification. Previously with usecallingpres=no everything was Ok, one small difference between the BRI (B410p) was that the callerid needed to be stripped from one number. But then came the need to make hidden calls, and so to enable usecallingpres and SetCallerPres(). ... Well, it was solved by changing prilocaldialplan=dynamic to prilocaldialplan=unknown Is there some good technical documentation about PRI lines somewhere ? there is a lot of voodoo magic in configuring thoses lines with asterisk ... -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: June 16, 2008 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! On Sun, 2008-06-15 at 18:04 +0200, randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! By tweaker, I assume you mean the small screwdrivers we often give away. I don't go anywhere without one (and I've even found that airport security tends not to take them away from me if I separate the plastic handle from the metal piece). I'm glad we were able to save your bacon. :-) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: June 16, 2008 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! On Sun, 2008-06-15 at 18:04 +0200, randulo wrote: Moving day, everything packed. Including tools! But wait, there in the jar with pens and pencils... it looks like. Yes, it's the Digium Asterisk tweaker! THANKS Digium! By tweaker, I assume you mean the small screwdrivers we often give away. I don't go anywhere without one (and I've even found that airport security tends not to take them away from me if I separate the plastic handle from the metal piece). I'm glad we were able to save your bacon. :-) LOL, good question...I wouldn't mind havin' one...i can haz tweaker? ROFL -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? :( I was thinking the same thing, Ottawa here.. :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need a SIP trunk provider for US - Dallas/TX
read the fine prints with them. Sometimes they have a thresh hold. Where you cannot exceed 1500 minutes per month on the trunk. I use a T1 which is cheaper if you compare it. On Fri, Jun 13, 2008 at 4:52 PM, Jonn R Taylor [EMAIL PROTECTED] wrote: I use bandwidth.com, works very well. 5 trunks start at about $125 a month. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of D. Dante Lorenso Sent: Friday, June 13, 2008 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Need a SIP trunk provider for US - Dallas/TX All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can guarantee $500 minimum monthly spending. Until business ramps up, I doubt I'll be doing that kind of spending immediately. Suggestions? -- Dante ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
I've been playing around in order to find something new and I've found this: I have created an IVR for test purposes, then I've placed a call from my sip phone using one of my telco lines to another of my telco lines attached to the PBX, in this situation I'm using two FXO channels, one for the outgoing call and another for the incoming call. Then I have created an extension in this IVR in order to make an echo test and I've used MixMonitor() to record the audio of the test. When I dial this extension I never can hear my echoed voice, but when I listen to the recording the audio have a lot of artifacts and the busy and dial tone are almost inaudible, the same effect that happens when you play to almost identical audio files, so I can presume that it is the same audio wave but out of phase (meaning the echo is working, I think). I don't know if this can be happening because of the Hardware Echo Canceler on my Remora A400D. If I call the extension of the echo test directly from my SIP phone without using any telco line (SIP -- IP -- Asterisk) then the test works just fine. Another test I've made is, during a call with the one way audio problem, I have used the ZapBarge() application to hear what's happening on the Zap Channel (from another SIP phone on my network). In this case I heard the callee complaining that he/she can't hear anything and I can't hear the caller (which is on the same network of my phone). In this case the caller can hear the callee. I have grabbed the sip debug messages of this call from the asterisk CLI and is attached (compressed) to this email. Well, thanks again for any comment/response... -- Nacho Linux Counter #156439 On Tue, Jun 17, 2008 at 5:14 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi Steve and the rest of the list, On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T My Asterisk Server has two NIC with a channel bonding setup (Balance TLB) connected to the same switch, and it does not have any firewall rule. I'm attaching a file with the output of sip set debug on the CLI of a call in this situation. Although calls made with SIP phones have this strange behavior, when I place a call with an analog phone connected to a FXS port of the same TDM card (see below for full description) this does not happen. Thanks, any help will be really appreciated... -- Nacho Linux Counter #156439 On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users SIP-Debug-141.txt.gz Description: GNU Zip compressed data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Center
Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Hello, We have set up dozens of call centers, some using Asterisk Queues and the rest using VICIDIAL Call Center Suite. What you want can easily be accomplished with an average server and Asterisk Queues with not too much effort using standard Asterisk configuration features. we have set up a small 7-seat inbound call center like this for a client on a P4 1.6GHz PC and it has worked great for the last 3 years. Thanks, MATT--- On 6/16/08, Sherwood McGowan [EMAIL PROTECTED] wrote: broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Re: OT How Digium Saved My Bacon!
Mark Hamilton wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! I hope JT is taking notes and will get the higher ups to add 'tweakers' to the digium store. G On a personal note, i still haven't seen my 'sticker'! haha ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
On Mon, Jun 16, 2008 at 8:24 PM, broadband Voice [EMAIL PROTECTED] wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. Read the book. You want to especially read about gotoiftime and queues. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Dear Sherwood, I am also using Asterisk Call Center Setup in my office with voice recording. The only thing I am unable to setup is web based call recording (CDR) access. From your email I think you have configured such a thing can you please share with me how can I also setup this solution. I know how to run and install Apache. Don't know abt PostgreSQL. However can do it if you can define some steps. And also how to integrate this all PostgreSQl+Apache+Web Based Links to Recordings. It will be a great help. regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, June 17, 2008 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Syed Nasruddin wrote:
Dear Sherwood,
I am also using Asterisk Call Center Setup in my office with voice
recording. The only thing I am unable to setup is web based call
recording (CDR) access. From your email I think you have configured such
a thing can you please share with me how can I also setup this solution.
I know how to run and install Apache. Don't know abt PostgreSQL. However
can do it if you can define some steps.
And also how to integrate this all PostgreSQl+Apache+Web Based Links to
Recordings. It will be a great help.
regards
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
broadband Voice wrote:
Is anyone using Asterisk as a call center. I want to be able to set it
up for my office line, when calls come in after 7:00pm Est want a
recording to says the office is closed and have about 5 phones that I
want to use as an agent. Can anyone share their implementation?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
There's a ton of us on here who have installations in call centers. What
would you like to know?
I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM
running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO
channels in a Rhino modular channel bank (all on the Digium card), and 2
24 port adtran total access channel banks running on the Tormenta. The
Adtrans drive the 40 analog phones for the sales floor, and we have 25
SIP phones. All phone conversations are recording by Asterisk and are
converted from GSM to Speex post-call by speexenc. We also run
PostgreSQL and Apache on the same system to serve CDRs with links to
recordings.
Anything else you'd like to know?
Syed,
What I did for a quick and dirty solution was install asterisk-stats and
modify the source code to include a link to the unique filename of the
recording (I use ${UNIQUEID}). This has worked just fine for our 75 or
so phone setup :) IIRC we found asterisk-stats on voip-info.org. We
just used that instead of creating an in house CDR web app, since the
client just needed a basic interface to look up calls and pull the
recordings.
If you'd like more information just let me know.
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Syed Nasruddin wrote: Dear Sherwood, I am also using Asterisk Call Center Setup in my office with voice recording. The only thing I am unable to setup is web based call recording (CDR) access. From your email I think you have configured such a thing can you please share with me how can I also setup this solution. I know how to run and install Apache. Don't know abt PostgreSQL. However can do it if you can define some steps. And also how to integrate this all PostgreSQl+Apache+Web Based Links to Recordings. It will be a great help. regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, June 17, 2008 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? Also, I personally prefer MySQL not PostGreSQL, I just got stuck with an existing PostGreSQL install. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need ata suggestion
I'm presently working on provisioning VoIP to a traditional key system. I have a single SIP DID inbound that gives me a maximum of 2 concurrent channels. I need an ATA that will ring the second station port when the first is in use. What devices will do this with a single sip registration with the provider? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Mon, Jun 16, 2008 at 11:11:00AM -0400, Steve Totaro wrote: On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote: Is there a contradiction between them? Naw; Steve's just showin' his ass again. Nah, just showing various tactics, sure some contradict each other. Yes, but clearly, neither Alex nor I thought that the two you quoted actually *do* contradict one another. He was just being polite. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Astricon question: four or five tracks?
On Sat, Jun 14, 2008 at 06:34:26PM -0700, John Todd wrote: I did receive one objection so far to the fifth track because of the fear of missing a good talk in the other four tracks. However, I'm not sure how we overcome that problem if we don't add the track, as we would have to dump 11 talks if we didn't have that additional room for speakers, meaning that the results would be identical: some good talks would not be able to be seen. In fact, no, John, it's worse: if you schedule the 5th track, *some* percentage of the attendees may find a conflict, and have to miss a presentation. If you *don't* schedule the 5th track, *100%* of the attendees will not see any of those 11 speakers. The suggestion by you and others of creating videos of the talks is a good one, and we had already started the discovery process of determining cost for doing just that. If we determine that it is within the budget, our goal would be to make the videos available to conference attendees as soon as possible after the talks are over to allow participants to see the talks that they had missed due to scheduling. This decision and arrangement is still pending, but we're hoping that we can do this to address the concerns of overlapping tracks. More on this hopefully soon. I would suspect that most (but possibly not all) of the videos would be available only to attendees if such a videography can be arranged. If your agreements with the speakers permit, you should probably not restrict sales to registered attendees; as I noted before, the reports I hear from conference arrangers seems to suggest that the videos won't cannibalize the show; hell, NANOG streams them, *and* gives the videos away for free... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Mon, Jun 16, 2008 at 09:30:12PM -0400, Matt Florell wrote: Any chance of more of these being handed out at Astricon this year? Oh, sure... beat me to the idea. :-) BTW: did the annotated syllabus notes from the Boot Camp ever finally get finished? I never got a copy. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reg call recording
Hi all I am using asterisk as pbx for my company. My company has requirement that all the incoming and outgoing calls should be recorded for all the extensions and should be able to play recorded call on extensions basis, that is , say 123 extension has made what call on the particular date and should be able to play and listen to it. What is the better way to achieve this? Any kind of suggestion is truly appreciated. Bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Thanks all for your responses. I will look into the Asterisk Queues and VICIDIAL Call Center Suite. On Mon, Jun 16, 2008 at 11:44 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Dear Sherwood, I am also using Asterisk Call Center Setup in my office with voice recording. The only thing I am unable to setup is web based call recording (CDR) access. From your email I think you have configured such a thing can you please share with me how can I also setup this solution. I know how to run and install Apache. Don't know abt PostgreSQL. However can do it if you can define some steps. And also how to integrate this all PostgreSQl+Apache+Web Based Links to Recordings. It will be a great help. regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, June 17, 2008 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? Also, I personally prefer MySQL not PostGreSQL, I just got stuck with an existing PostGreSQL install. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg call recording
Bikrish Amatya wrote:
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded call on extensions basis,
that is , say 123 extension has made what call on the particular date
and should be able to play and listen to it. What is the better way to
achieve this? Any kind of suggestion is truly appreciated.
Bikrish
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
A simple web interface, such as asterisk-stats coupled with some basic
modifications to link to a recording that was made with ${UNIQUEID} as
the recording filename (pre extension, use monitor + soxmix to mix the
recordings) will work just fine, I use it on a medium-large installation
that does about 10K calls a day, with no issues in regards to recordings
or ability to access calls/recordings.
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Mon, 16 Jun 2008, Eric Fort wrote: I'm presently working on provisioning VoIP to a traditional key system. I have a single SIP DID inbound that gives me a maximum of 2 concurrent channels. I need an ATA that will ring the second station port when the first is in use. What devices will do this with a single sip registration with the provider? Er, Asterisk will do this with a TDM400 card or clone. Expensive ATA though :) But maybe an AVM Fritz! box will work for you too... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: OT How Digium Saved My Bacon!
The screwdriver is reversible, it swings both ways, pull out the shank and stick it in the other way, it becomes a Phillips. I'm tellin ya, there Digium engineers are good! (Phillips is a trademark of Phillips) On Tue, Jun 17, 2008 at 3:28 AM, Richard Lyman [EMAIL PROTECTED] wrote: Mark Hamilton wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! I hope JT is taking notes and will get the higher ups to add 'tweakers' to the digium store. G On a personal note, i still haven't seen my 'sticker'! haha ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 1:22 AM, Mark Hamilton [EMAIL PROTECTED] wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? :( There was an Astricon in Paris. After my presentation, I was awarded the get screwed gift pack! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Mon, Jun 16, 2008 at 10:43 PM, Jared Smith [EMAIL PROTECTED] wrote: By tweaker, I assume you mean the small screwdrivers we often give Actually, I think I overstepped the true definition of tweaker. This is a real screwdriver with anti-geekdom feature. It looks like a pen when it is stuck in a jar with pens. A tweaker is a lot smaller and less versatile. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
