Re: [asterisk-users] 2000+ user Asterisk PBX
On Sat, Aug 2, 2008 at 5:45 PM, Femi [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Rowson Sent: 02 August 2008 19:42 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 2000+ user Asterisk PBX Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs Femi I would be interested in some of the replies if you wanted to continue the topic on-list... Your problem might help someone else down the line. Me too, Any reason you want this off the list particularly? Chris Sorry if I appear selfish by asking for it to be off list Please by all means post non-commercial replies to my request on-list however any responses of a commercial nature (and that is primarily what I'm looking for) would naturally be off-list. Femi Yes there are. I am putting one together currently and have done 1,000 in the past, it is just a matter of putting the right things in the right places. Obviously, a decent sized budget is required, some TLC, and a few racks if you want good redundancy. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI device is down
Hello, My Digium wct4xxp suddenly stopped working. Here are some of the logs: zap restart [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:7164 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:10471 build_channels: Unable to register channel '1-15' [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:9768 zap_restart: Reload channels from zap config failed! I can supply other logs, but what can be wrong with the card/system? Thanks, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
On Sat, Aug 2, 2008 at 2:53 PM, Femi [EMAIL PROTECTED] wrote: Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs If you're talking about 2000+ SIP users then I have some experience. When we got to 1000 users we replaced the Digium E1 cards with a Cisco AS5400 to make our echo and static problems go away (that was 3 years ago so the cards may have improved since then). Now we've largely replaced the Cisco AS5400 with an SS7 switch but that was more a business call related to supplier interconnect considerations than a technical one. Those Cisco AS5400's are fantastic pieces of kit, it's the only thing in our set up that we have never had even the smallest problem with in over 3 years! An AS5400 will be more expensive than a commodity server with 2 or 3 four port E1 cards which is the main drawback. At the time it was more expensive for us to be continuously troubleshooting static and echo issues and as mentioned above that situation may have improved. If you are talking SIP users the other thing you'll have to do is to move as much of the non-call related SIP traffic off Asterisk. That means a separate SIP Proxy and/or Registrar. We have split those two functions off onto separate boxes so that Asterisk only has to deal with SIP for call signalling and of course media which is what its big strength is. Finally you will need a nice big database box, realtime is the only practical way to run Asterisk for anything over a few hundred users. The SIP Registrar and Asterisk will both generate large loads on your database and it is the critical link in the chain. If your database has problems you can't get any calls out and if that's not bad enough all your user's ATAs registrations will drop off meaning you get a deluge of support calls. We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution which means the time it takes to switch over from the primary to secondary database is a problem. MySQL seems to have a much better replication solution and Asterisk doesn't really need a lot of the advantages Postgresql has over MySQL such as better stored procedure support etc. So in summary the critical factors in my opinion are: 1. Good database, 2. Good quality solution ofr E1's, 3. Split off non-media related signalling from Asterisk. If you're not talking about 2000+ SIP users then you can pretty much disregard everything I've said :-). Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lookup for '_sip._udp.sip.stanaphone.com'
On Sat, Aug 02, 2008 at 11:30:54PM -0400, Dean Collins wrote: I am having problems with my sip service with stanaphone. I think it is related to my firewall which had a glitch yesterday. Can anyone tell me what this means below? -- ast_get_srv: SRV lookup for '_sip._udp.sip.stanaphone.com' mapped to host sip.stanaphone.com, port 5060 Aunt Wiki can. http://en.wikipedia.org/wiki/SRV_record -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI device is down
On Sun, Aug 03, 2008 at 10:05:46AM +0300, Elliot Murdock wrote: Hello, My Digium wct4xxp suddenly stopped working. Here are some of the logs: zap restart [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:7164 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:10471 build_channels: Unable to register channel '1-15' [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:9768 zap_restart: Reload channels from zap config failed! I can supply other logs, but what can be wrong with the card/system? Can you try instead: 'module unload chan_zap.so' and then: 'module load chan_zap.so' ? If that doesn't work, what is the output of: cat /proc/zaptel/* -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI device is down
Thanks Tzafrir, This is what I get for various logs: cat /proc/interrupts CPU0 CPU1 0: 900209 0 XT-PIC timer 2: 0 0 XT-PIC cascade 5:263 0 XT-PIC uhci_hcd:usb4, HDA Intel, [EMAIL PROTECTED]::00:02.0 6: 2 0 XT-PIC uhci_hcd:usb1, ehci_hcd:usb5 7: 1 0 XT-PIC parport0 8: 50 0 XT-PIC rtc 9: 8 0 XT-PIC acpi, wct4xxp 10: 112574 0 XT-PIC eth0 11: 0 0 XT-PIC uhci_hcd:usb2, uhci_hcd:usb3 14: 24645 0 XT-PIC libata 15: 25758 0 XT-PIC libata NMI: 0 0 LOC: 899964 899835 ERR: 0 MIS: 0 module unload chan_zap.so -- Unregistered channel -2 -- Unregistered channel 1 ... -- Unregistered channel 122 -- Unregistered channel 123 -- Unregistered channel 124 CLI module load chan_zap.so [Aug 3 11:35:40] ERROR[5518]: chan_zap.c:9415 start_pri: Unable to open D-channel 16 (Device or resource busy) [Aug 3 11:35:40] ERROR[5518]: chan_zap.c:11327 setup_zap: Unable to start D-channel on span 1 This is the output of /proc/zaptel/1: Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 ClockSource 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) 3 TE4/0/1/3 Clear (In use) 4 TE4/0/1/4 Clear (In use) 5 TE4/0/1/5 Clear (In use) 6 TE4/0/1/6 Clear (In use) 7 TE4/0/1/7 Clear (In use) 8 TE4/0/1/8 Clear (In use) 9 TE4/0/1/9 Clear (In use) 10 TE4/0/1/10 Clear (In use) 11 TE4/0/1/11 Clear (In use) 12 TE4/0/1/12 Clear (In use) 13 TE4/0/1/13 Clear (In use) 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use) 18 TE4/0/1/18 Clear (In use) 19 TE4/0/1/19 Clear (In use) 20 TE4/0/1/20 Clear (In use) 21 TE4/0/1/21 Clear (In use) 22 TE4/0/1/22 Clear (In use) 23 TE4/0/1/23 Clear (In use) 24 TE4/0/1/24 Clear (In use) 25 TE4/0/1/25 Clear (In use) 26 TE4/0/1/26 Clear (In use) 27 TE4/0/1/27 Clear (In use) 28 TE4/0/1/28 Clear (In use) 29 TE4/0/1/29 Clear (In use) 30 TE4/0/1/30 Clear (In use) 31 TE4/0/1/31 Clear (In use) Span 2, 3, and 4 are the same except the span number. pri show spans gives me: s span www*CLI pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned, Down, Active PRI span 4/0: Provisioned, Down, Active Thank you, Elliot On 8/3/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Aug 03, 2008 at 11:25:33AM +0300, Tzafrir Cohen wrote: On Sun, Aug 03, 2008 at 10:05:46AM +0300, Elliot Murdock wrote: Hello, My Digium wct4xxp suddenly stopped working. Here are some of the logs: zap restart [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:7164 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:10471 build_channels: Unable to register channel '1-15' [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:9768 zap_restart: Reload channels from zap config failed! I can supply other logs, but what can be wrong with the card/system? Can you try instead: 'module unload chan_zap.so' and then: 'module load chan_zap.so' ? If that doesn't work, what is the output of: cat /proc/zaptel/* err... misread your message as wctdm24xxp instead of wct4xxp. Try restarting asterisk of that 'unload' and 'load' doesn't work. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI device is down
Thanks Tzafrir, This is what I get: module unload chan_zap.so -- Unregistered channel -2 -- Unregistered channel 1 ... -- Unregistered channel 122 -- Unregistered channel 123 -- Unregistered channel 124 CLI module load chan_zap.so [Aug 3 11:35:40] ERROR[5518]: chan_zap.c:9415 start_pri: Unable to open D-channel 16 (Device or resource busy) [Aug 3 11:35:40] ERROR[5518]: chan_zap.c:11327 setup_zap: Unable to start D-channel on span 1 This is the output of /proc/zaptel/1: Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 ClockSource 1 TE4/0/1/1 Clear (In use) 2 TE4/0/1/2 Clear (In use) 3 TE4/0/1/3 Clear (In use) 4 TE4/0/1/4 Clear (In use) 5 TE4/0/1/5 Clear (In use) 6 TE4/0/1/6 Clear (In use) 7 TE4/0/1/7 Clear (In use) 8 TE4/0/1/8 Clear (In use) 9 TE4/0/1/9 Clear (In use) 10 TE4/0/1/10 Clear (In use) 11 TE4/0/1/11 Clear (In use) 12 TE4/0/1/12 Clear (In use) 13 TE4/0/1/13 Clear (In use) 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use) 16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use) 18 TE4/0/1/18 Clear (In use) 19 TE4/0/1/19 Clear (In use) 20 TE4/0/1/20 Clear (In use) 21 TE4/0/1/21 Clear (In use) 22 TE4/0/1/22 Clear (In use) 23 TE4/0/1/23 Clear (In use) 24 TE4/0/1/24 Clear (In use) 25 TE4/0/1/25 Clear (In use) 26 TE4/0/1/26 Clear (In use) 27 TE4/0/1/27 Clear (In use) 28 TE4/0/1/28 Clear (In use) 29 TE4/0/1/29 Clear (In use) 30 TE4/0/1/30 Clear (In use) 31 TE4/0/1/31 Clear (In use) Span 2, 3, and 4 are the same except the span number. pri show spans gives me: s span www*CLI pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned, Down, Active PRI span 4/0: Provisioned, Down, Active Thank you, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI device is down
On Sun, Aug 03, 2008 at 11:25:33AM +0300, Tzafrir Cohen wrote: On Sun, Aug 03, 2008 at 10:05:46AM +0300, Elliot Murdock wrote: Hello, My Digium wct4xxp suddenly stopped working. Here are some of the logs: zap restart [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:7164 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Aug 3 10:02:55] ERROR[15050]: chan_zap.c:10471 build_channels: Unable to register channel '1-15' [Aug 3 10:02:55] WARNING[15050]: chan_zap.c:9768 zap_restart: Reload channels from zap config failed! I can supply other logs, but what can be wrong with the card/system? Can you try instead: 'module unload chan_zap.so' and then: 'module load chan_zap.so' ? If that doesn't work, what is the output of: cat /proc/zaptel/* err... misread your message as wctdm24xxp instead of wct4xxp. Try restarting asterisk of that 'unload' and 'load' doesn't work. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Registration
Hi, I have this weird problem i cant explain. i have two asterisk, i'm using realtime table for my sip/user accounts. my database is on a mysql cluster. my prob is if i register on phone on asterisk 1 it is ok, but on second asterisk it can't, Registration from '122144 sip:[EMAIL PROTECTED]:5060' failed for '12.34.56.78' - Wrong password but both asterisk talks to a single mysql cluster. i defined this on my sip.conf domain=10.10.10.130 domain=10.10.10.131 domain=my.domain.com any ides? TIA Regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Least Cost Routing
Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No MOH on SIP hold nor on park
Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to no. Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using # and when using the hold functions of my Cisco 7960 (SCCP), music on hold works without problems. I'm running Asterisk 1.4.21.1. IIRC, MOH on parked calls was working earlier but I didn't use the park functions extensively so I don't remember exactly when that was. Any ideas? --Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I issue a Flash to Zap (PSTN) from SIP?
Cool, thanks for the tip. Why do you need to separate incoming and outgoing? Jim C F wrote: This is what I do: /etc/asterisk/features.conf [applicationmap] inflash = *4,caller,Flash,() outflash = *3,callee,Flash,() in /etc/asterisk/extensions.conf before accepting a call: exten = s,n,Set(DYNAMIC_FEATURES=inflash) on an outgoing call: exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash) in incoming calls the user has to press *4 on outgoing calls the user has to press *3 On Sat, Aug 2, 2008 at 4:59 PM, Jim Duda [EMAIL PROTECTED] wrote: I've seen a few posts on this issue, however, no definitive answer. My PSTN is connected to Zap/4. I have simple Call Waiting service on the PSTN line. All the other phones are SIP clients. When I'm on an Zap/SIP connection and another call comes in, I can hear the Call Waiting Tone on the SIP line. How can I issue a Flash/Hook to the Zap line in order to accept the other call? Also, is there any means to get Caller ID for the other call? I've seen posts that I can use *0 or *3 to send the Flash/Hook, however, that doesn't work for me. I realize there is a Flash( ) Dialplan function. How can I use this Function in the Dialplan with a call which is currently in progress? Any advice is most appreciated. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
On Sun, Aug 3, 2008 at 3:13 AM, Grey Man [EMAIL PROTECTED] wrote: On Sat, Aug 2, 2008 at 2:53 PM, Femi [EMAIL PROTECTED] wrote: Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs If you're talking about 2000+ SIP users then I have some experience. When we got to 1000 users we replaced the Digium E1 cards with a Cisco AS5400 to make our echo and static problems go away (that was 3 years ago so the cards may have improved since then). Now we've largely replaced the Cisco AS5400 with an SS7 switch but that was more a business call related to supplier interconnect considerations than a technical one. Those Cisco AS5400's are fantastic pieces of kit, it's the only thing in our set up that we have never had even the smallest problem with in over 3 years! An AS5400 will be more expensive than a commodity server with 2 or 3 four port E1 cards which is the main drawback. At the time it was more expensive for us to be continuously troubleshooting static and echo issues and as mentioned above that situation may have improved. If you are talking SIP users the other thing you'll have to do is to move as much of the non-call related SIP traffic off Asterisk. That means a separate SIP Proxy and/or Registrar. We have split those two functions off onto separate boxes so that Asterisk only has to deal with SIP for call signalling and of course media which is what its big strength is. Finally you will need a nice big database box, realtime is the only practical way to run Asterisk for anything over a few hundred users. The SIP Registrar and Asterisk will both generate large loads on your database and it is the critical link in the chain. If your database has problems you can't get any calls out and if that's not bad enough all your user's ATAs registrations will drop off meaning you get a deluge of support calls. We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution which means the time it takes to switch over from the primary to secondary database is a problem. MySQL seems to have a much better replication solution and Asterisk doesn't really need a lot of the advantages Postgresql has over MySQL such as better stored procedure support etc. So in summary the critical factors in my opinion are: 1. Good database, 2. Good quality solution ofr E1's, 3. Split off non-media related signalling from Asterisk. If you're not talking about 2000+ SIP users then you can pretty much disregard everything I've said :-). Regards, Greyman. Curious why you stay with postgres then, and not go with MySQL if you know in advance it is a problem and will bite you sometime? You way want to look at extconig.conf and ODBC or whatever database driver and even hook up to a MSSQL cluster. While not inexpensive, it is is mission critical. If you can monetize one hour of downtime and then figure the MTBF (including all the pieces) and how long ot trouble shoot and get it back up, maybe that is worth quite a bit of money. If you can just say sorry, we were upgrading and the cost is only time, then that certainly dictates your direction. As for the Digium cards, they are WAY better now, I think a little competition in the regards to cards from Sangoma made up for that. Anyways, most providers are offering SIP over IP nowdays or starting to. I know GXing can usually supply a point to point to their PSTN, DS3/T3 or whatever, then with g729 you can get alot of calls across that pipe. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDMoE with Telco
Hello, is it possible with TDMoE to replace classic digital T1/E1 interfaces like digium and sangoma cards connected to a telco. Or TDMoE is only possible for connecting two asterisk boxes using their NIC interfaces. if TDMoE can work with an T1/E1 connected with telco how we can get the remote mac address of the telco interface ? Thanks___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
Thanks Steve and Greyman, Thanks for the design tips Based on all the information I have been able to gather this is the config that I believe will work best: 1. SER on 2 servers in HA (failover) config 2. Asterisk cluster of 4 (or more) servers 3. MySQL and SMTP on 2 servers with HA config 4. Cisco or FoneBridge E1 gateways for telco access 5. Other servers for ITSP access 6. All systems separated on two racks at different ends of the building 7. HA switches, routers and firewalls Now here's the reason why I need some support All of this somehow has to be tied up with a simple web based management console for the administrator and another for the operator Configuring new routes and user voicemail / email has to be seamless and from the same central console For the users I need fax to mail and voicemail to mail as well a web interface for voicemail This is where I believe the real pitfalls are as the hardware and platform issues are well documented Femi -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 03 August 2008 18:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2000+ user Asterisk PBX On Sun, Aug 3, 2008 at 3:13 AM, Grey Man [EMAIL PROTECTED] wrote: On Sat, Aug 2, 2008 at 2:53 PM, Femi [EMAIL PROTECTED] wrote: Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs If you're talking about 2000+ SIP users then I have some experience. When we got to 1000 users we replaced the Digium E1 cards with a Cisco AS5400 to make our echo and static problems go away (that was 3 years ago so the cards may have improved since then). Now we've largely replaced the Cisco AS5400 with an SS7 switch but that was more a business call related to supplier interconnect considerations than a technical one. Those Cisco AS5400's are fantastic pieces of kit, it's the only thing in our set up that we have never had even the smallest problem with in over 3 years! An AS5400 will be more expensive than a commodity server with 2 or 3 four port E1 cards which is the main drawback. At the time it was more expensive for us to be continuously troubleshooting static and echo issues and as mentioned above that situation may have improved. If you are talking SIP users the other thing you'll have to do is to move as much of the non-call related SIP traffic off Asterisk. That means a separate SIP Proxy and/or Registrar. We have split those two functions off onto separate boxes so that Asterisk only has to deal with SIP for call signalling and of course media which is what its big strength is. Finally you will need a nice big database box, realtime is the only practical way to run Asterisk for anything over a few hundred users. The SIP Registrar and Asterisk will both generate large loads on your database and it is the critical link in the chain. If your database has problems you can't get any calls out and if that's not bad enough all your user's ATAs registrations will drop off meaning you get a deluge of support calls. We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution which means the time it takes to switch over from the primary to secondary database is a problem. MySQL seems to have a much better replication solution and Asterisk doesn't really need a lot of the advantages Postgresql has over MySQL such as better stored procedure support etc. So in summary the critical factors in my opinion are: 1. Good database, 2. Good quality solution ofr E1's, 3. Split off non-media related signalling from Asterisk. If you're not talking about 2000+ SIP users then you can pretty much disregard everything I've said :-). Regards, Greyman. Curious why you stay with postgres then, and not go with MySQL if you know in advance it is a problem and will bite you sometime? You way want to look at extconig.conf and ODBC or whatever database driver and even hook up to a MSSQL cluster. While not inexpensive, it is is mission critical. If you can monetize one hour of downtime and then figure the MTBF (including all the pieces) and how long ot trouble shoot and get it back up, maybe that is worth quite a bit of money. If you can just say sorry, we were upgrading and the cost is only time, then that certainly dictates your direction. As for the Digium cards, they are WAY better now, I think a little competition in the regards to cards from Sangoma made up for that. Anyways, most providers are offering SIP over IP nowdays or starting to. I know GXing can usually supply a point to point to their PSTN, DS3/T3 or whatever, then with g729 you can get alot of calls across that pipe.
Re: [asterisk-users] 2000+ user Asterisk PBX
On Sun, Aug 03, 2008 at 08:13:30AM +0100, Grey Man wrote: We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution Hmmm... is that really the case? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
Question, why did you not just cut to the chase. It is obvious you already knew the design. Why not pose the question to the biz list, if anyone could custom program this for you? Thanks, Steve Totaro On Sun, Aug 3, 2008 at 2:11 PM, Femi [EMAIL PROTECTED] wrote: Thanks Steve and Greyman, Thanks for the design tips Based on all the information I have been able to gather this is the config that I believe will work best: 1. SER on 2 servers in HA (failover) config 2. Asterisk cluster of 4 (or more) servers 3. MySQL and SMTP on 2 servers with HA config 4. Cisco or FoneBridge E1 gateways for telco access 5. Other servers for ITSP access 6. All systems separated on two racks at different ends of the building 7. HA switches, routers and firewalls Now here's the reason why I need some support All of this somehow has to be tied up with a simple web based management console for the administrator and another for the operator Configuring new routes and user voicemail / email has to be seamless and from the same central console For the users I need fax to mail and voicemail to mail as well a web interface for voicemail This is where I believe the real pitfalls are as the hardware and platform issues are well documented Femi -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 03 August 2008 18:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2000+ user Asterisk PBX On Sun, Aug 3, 2008 at 3:13 AM, Grey Man [EMAIL PROTECTED] wrote: On Sat, Aug 2, 2008 at 2:53 PM, Femi [EMAIL PROTECTED] wrote: Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs If you're talking about 2000+ SIP users then I have some experience. When we got to 1000 users we replaced the Digium E1 cards with a Cisco AS5400 to make our echo and static problems go away (that was 3 years ago so the cards may have improved since then). Now we've largely replaced the Cisco AS5400 with an SS7 switch but that was more a business call related to supplier interconnect considerations than a technical one. Those Cisco AS5400's are fantastic pieces of kit, it's the only thing in our set up that we have never had even the smallest problem with in over 3 years! An AS5400 will be more expensive than a commodity server with 2 or 3 four port E1 cards which is the main drawback. At the time it was more expensive for us to be continuously troubleshooting static and echo issues and as mentioned above that situation may have improved. If you are talking SIP users the other thing you'll have to do is to move as much of the non-call related SIP traffic off Asterisk. That means a separate SIP Proxy and/or Registrar. We have split those two functions off onto separate boxes so that Asterisk only has to deal with SIP for call signalling and of course media which is what its big strength is. Finally you will need a nice big database box, realtime is the only practical way to run Asterisk for anything over a few hundred users. The SIP Registrar and Asterisk will both generate large loads on your database and it is the critical link in the chain. If your database has problems you can't get any calls out and if that's not bad enough all your user's ATAs registrations will drop off meaning you get a deluge of support calls. We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution which means the time it takes to switch over from the primary to secondary database is a problem. MySQL seems to have a much better replication solution and Asterisk doesn't really need a lot of the advantages Postgresql has over MySQL such as better stored procedure support etc. So in summary the critical factors in my opinion are: 1. Good database, 2. Good quality solution ofr E1's, 3. Split off non-media related signalling from Asterisk. If you're not talking about 2000+ SIP users then you can pretty much disregard everything I've said :-). Regards, Greyman. Curious why you stay with postgres then, and not go with MySQL if you know in advance it is a problem and will bite you sometime? You way want to look at extconig.conf and ODBC or whatever database driver and even hook up to a MSSQL cluster. While not inexpensive, it is is mission critical. If you can monetize one hour of downtime and then figure the MTBF (including all the pieces) and how long ot trouble shoot and get it back up, maybe that is worth quite a bit of money. If you can just say sorry, we were upgrading and the cost is only time, then that certainly dictates your direction. As for the Digium cards, they are WAY better now, I think a little
Re: [asterisk-users] TDMoE with Telco
--Original Message Text--- From: Yacine Boukaba Date: Sun, 3 Aug 2008 18:54:08 +0100 Hello, is it possible with TDMoE to replace classic digital T1/E1 interfaces like digium and sangoma cards connected to a telco. Or TDMoE is only possible for connecting two asterisk boxes using their NIC interfaces. if TDMoE can work with an T1/E1 connected with telco how we can get the remote mac address of the telco interface ? ThanksNo virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 8/1/2008 6:59 PM I thought that TDMoE was largely depricated in the wake of DUNDi? Michael -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDMoE with Telco
On Sun, Aug 3, 2008 at 2:32 PM, Michael Graves [EMAIL PROTECTED] wrote: --Original Message Text--- From: Yacine Boukaba Date: Sun, 3 Aug 2008 18:54:08 +0100 Hello, is it possible with TDMoE to replace classic digital T1/E1 interfaces like digium and sangoma cards connected to a telco. Or TDMoE is only possible for connecting two asterisk boxes using their NIC interfaces. if TDMoE can work with an T1/E1 connected with telco how we can get the remote mac address of the telco interface ? ThanksNo virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 8/1/2008 6:59 PM I thought that TDMoE was largely depricated in the wake of DUNDi? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] Not as far as HA Asterisk and PRIs using products such as the Redfone's fonebridge. To the original poster, I seriously doubt it, never heard of anyone doing this and ANY network issues are going to ruin your calls. I think your best bet would be to find an ITSP, preferably that handles both the IP and PSTN sides of the equation, then you could utilize G729 and get more calls out of the pipe. I guess the real question is, why are you asking this question? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Random reboots on IP-601 after changing network topology
Hi, We did move an office from a remote building to another floor in our building, allowing us to directly hook the switch in that building to our core switch (GigE). In the past, the phones were on the same subnet as the * server. All the phones worked flawlessy for about two years. Since the move, the phones are on the network mysubnet.13.0/24 and the * server on mysubnet.0.0/24. Randomly my IP601 (430's are not affected by this bug) are rebooting. I've got the boot app log of the phones should it help to track the problem down. The phones are all hooked to a 3COM 4500 PWR switch. If i move the IP601 to the same VLAN as the * server, the phones are not rebooting anymore. Did someone already experience such a behavior ? Thanks Laurent 0803215603|copy |4|03|Upload of 'log/0004f2187952-app.log' FAILED on attempt 1 (addr 1 of 1) 0803215603|copy |4|03|Upload of 'log/0004f2187952-app.log' FAILED on attempt 1 (addr 1 of 1) 0803195742|so |*|03|-- Initial log entry -- 0803195742|so |*|03|Platform: Model=SoundPoint IP 601, Assembly=2345-11605-001 Rev=B 0803195742|so |*|03|Platform: MAC=0004f2187952, IP=mysubnet.13.195, Subnet Mask=255.255.255.0 0803195742|so |*|03|Platform: BootBlock=2.6.0 (11605_001) 30-Apr-05 12:50 0803195742|so |*|03|Platform: Bootrom=4.1.1.0232 29-Mar-08 16:39 0803195742|so |*|03|Application, main: Label=SIP, Version=3.0.3.0401 22-May-08 15:13 0803195742|so |*|03|Application, main: P/N=3150-11530-303 0803195742|ethf |*|03|Initial log entry. Current logging level 4 0803195742|so |5|03|utilCertificateInit failed. 0803195742|hw |*|03|Initial log entry. Current logging level 4 0803195742|ares |*|03|Initial log entry. Current logging level 4 0803195742|dns |*|03|Initial log entry. Current logging level 3 0803195742|cfg |*|03|Initial log entry. Current logging level 3 0803195742|cfg |3|03|RT|Checking DHCP option 160 type string 0803195742|cfg |3|03|RT|Runtime basic IP parameters updated. 0803195742|cfg |3|03|RT|Runtime provisioning server parameters updated. 0803195742|cfg |3|03|RT|Runtime SNTP parameters updated. 0803195742|dns |*|03|DNS resolver servers are 'mysubnet.0.3' 'mysubnet.0.2' 0803195742|dns |*|03|DNS resolver search domain is 'mydomain.com' 0803195742|log |*|03|Initial log entry. Current logging level 4 0803195742|so |4|03|[SoFontsC]: Font item (6)(1) is NULL. 0803195742|curl |*|03|Initial log entry. Current logging level 3 0803195742|utilm|*|03|Initial log entry. Current logging level 4 0803195742|copy |*|03|Initial log entry. Current logging level 3 0803195742|rtos |*|03|Initial log entry. Current logging level 4 0803195742|sec |*|03|Initial log entry. Current logging level 4 0803195742|cfg |3|03|Prm|Beginning to provision phone 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/2345-11605-001.bootrom.ld' from 'mysubnet.0.3' 0803195742|cfg |3|03|Prm|Image 2345-11605-001.bootrom.ld has not changed 0803195742|copy |3|03|buffered_write: transfer Terminated on entry. Return 0 0803195742|copy |3|03|Download of '2345-11605-001.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0803195742|cfg |3|03|Prm|Downloaded bootROM is identical to current version 4.1.1 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/0004f2187952.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of '0004f2187952.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/2345-11605-001.sip.ld' from 'mysubnet.0.3' 0803195742|cfg |3|03|Prm|Image 2345-11605-001.sip.ld has not changed 0803195742|copy |3|03|buffered_write: transfer Terminated on entry. Return 0 0803195742|copy |3|03|Download of '2345-11605-001.sip.ld' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/phone-0156800677.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of 'phone-0156800677.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|File /ffs0/phone-0156800677_cfg.zzz, is upto date 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/sip.cfg' from 'mysubnet.0.3' 0803195742|copy |3|03|Download of 'sip.cfg' succeeded on attempt 1 (addr 1 of 1) 0803195742|copy |3|03|File /ffs0/sip_cfg.zzz, is upto date 0803195742|cfg |3|03|Prm|Check of configuration files suceeded 0803195742|cfg |3|03|Prm|Phone successfully provisioned 0803195742|cfg |*|03|Prm|Configuration file phone-0156800677.cfg is from template phone1.cfg, revision 1.83.2.2 0803195742|cfg |*|03|Prm|Configuration file sip.cfg is from template sip.cfg, revision 1.273.2.69 0803195742|so |*|03|Configuration files: phone-0156800677.cfg,sip.cfg 0803195742|copy |3|03|'ftp://PlcmSpIp:[EMAIL PROTECTED]/0004f2187952-phone.cfg' from 'mysubnet.0.3' 0803195742|copy |4|03|Download of '0004f2187952-phone.cfg' FAILED on attempt 1 (addr 1 of 1) 0803195742|copy |4|03|Server 'mysubnet.0.3' said '0004f2187952-phone.cfg' is not present 0803195742|utilm|4|03|uBLFCompressed: File /ffs0/local/0004f2187952-phone_cfg.zzz doesn't
Re: [asterisk-users] how many quad T1 cards
Tilghman Lesher [EMAIL PROTECTED] writes: I'm not terribly sure that the PCI bus will stand up to that many interrupts per second, though it's certainly possible. The PCI bus should be rather bored with 2Mbps per card. Only one card should interrupt, but I am not sure whether Sangoma or Digium cards are clever enough to do that. (In an ideal world they wouldn't interrupt at all, the driver would poll.) Last I heard the PCI bus was nearly at capacity servicing just 3 quad-span cards (note that the PCI bus has other things to service, like hard drive accesses, network, keyboard, etc.). With PCI-Express every card has dedicated bandwidth. You'll probably do better with two machines, rather than trying to stack everything into one. Now that I agree with. It would probably take serious optimizations to get everything working. It would be fun to try actually. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDMoE with Telco
You can use TDMoE to get an E1 running but its really designed to replicate an E1 end to end Its a standard and there is equipment out there that does it, e.g. from RAD and a few others. I didn't have any joy using the Asterisk code to get it going but it should in theory work. Its completely different to Dundi The challenge it is a protocol and needs two boxes talking TDMoE at each end. Telco's do not have this as an option, or at least none do that I have found Cheers Duncan Michael Graves wrote: --Original Message Text--- *From:* Yacine Boukaba *Date:* Sun, 3 Aug 2008 18:54:08 +0100 Hello, is it possible with TDMoE to replace classic digital T1/E1 interfaces like digium and sangoma cards connected to a telco. Or TDMoE is only possible for connecting two asterisk boxes using their NIC interfaces. if TDMoE can work with an T1/E1 connected with telco how we can get the remote mac address of the telco interface ? ThanksNo virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 8/1/2008 6:59 PM I thought that TDMoE was largely depricated in the wake of DUNDi? Michael -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDMoE with Telco
-Original Message- Not as far as HA Asterisk and PRIs using products such as the Redfone's fonebridge. To the original poster, I seriously doubt it, never heard of anyone doing this and ANY network issues are going to ruin your calls. I think your best bet would be to find an ITSP, preferably that handles both the IP and PSTN sides of the equation, then you could utilize G729 and get more calls out of the pipe. I guess the real question is, why are you asking this question? Thanks, Steve T My guess is what the original poster wants to know is if boxes like the redFone exist that allow you to set up your Asterisk box without having to directly plug in TDM cards like those from Digium and Sangoma. The short answer to this question is yes. There are a few solutions that run on TDMoE like the PhoneBridge redFone that plug into the T1/E1 from the telco on one end and connect to the Asterisk box via IP. Now about the MAC address, if you are talking about the MAC address of the TDMoE box that can easily be obtained but the MAC address of the telco's E1?? Not sure what you mean. Regards, Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad recorded audio quality (upgrade).
Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old recorded VM prompts) sound HORRIBLE. Call quality is great, both internal and external. Any idea as to what might have happened? Could I have brought over a config that's not valid for this setup? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
I can speak first hand to this having gone through it just a few months ago . . After being spoiled with all the features and standard compliance in Postgres, I was put in a position with a new project to setup a redundant (Master-Slave) database cluster. I immediately jumped to Postgres to do the job (using 8.3). My biggest gripe at the time was that there was really nothing built IN postgres to do the replication as I soon found out. Everything was third party and there were several replication modules suggested to me that seemed stagnant or un-maintained or required an older version of Postgres (bypassing the massive performance increase of the 8.3 release). Of those that I did try that were opensource, all of them seemed fairly complex to get up and running - to say the least. Also having used MySQL extensively, I decided to give it a test run on a separate set of boxes. I'm not exaggerating when I say the replication was up and running in about 10 minutes. While I do appreciate (a lot) how standards compliant Postgres is, MySQL was an absolute clear winner in my book with regards to the replication. Just my two cents . . - Darren _ [EMAIL PROTECTED] http://www.darrensessions.com http://www.linkedin.com/in/dsessions _ On Aug 3, 2008, at 12:26 PM, Tzafrir Cohen wrote: On Sun, Aug 03, 2008 at 08:13:30AM +0100, Grey Man wrote: We use Postgresql which does a good job but the big problem with it is redundancy. Postgresql does not really have an industrial strength replication solution Hmmm... is that really the case? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
On 16:20, Sun 03 Aug 08, Darren Sessions wrote: I can speak first hand to this having gone through it just a few months ago . . After being spoiled with all the features and standard compliance in Postgres, I was put in a position with a new project to setup a redundant (Master-Slave) database cluster. I immediately jumped to Postgres to do the job (using 8.3). My biggest gripe at the time was that there was really nothing built IN postgres to do the replication as I soon found out. Everything was third party and there were several replication modules suggested to me that seemed stagnant or un-maintained or required an older version of Postgres (bypassing the massive performance increase of the 8.3 release). Of those that I did try that were opensource, all of them seemed fairly complex to get up and running - to say the least. Also having used MySQL extensively, I decided to give it a test run on a separate set of boxes. I'm not exaggerating when I say the replication was up and running in about 10 minutes. While I do appreciate (a lot) how standards compliant Postgres is, MySQL was an absolute clear winner in my book with regards to the replication. Amen. been there and been bitten by the same stuff. We are now using a 4 node mysql master-master setup which works great. Ok, the total setuptime was closer to an hour then two minutes, but that's because we wanted write access to all nodes. make sure to setup the primary key start and increment config params correctly, and you're done. Just my two cents . . My two cents and two weeks of investigation+testing+redoing_it_over_and_over_again - Darren Hmmm... is that really the case? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 6
Salve, dal quattro all'otto Agosto non saro' in ufficio. In mia assenza il referente in ufficio e' l'ing. Maurizio Intravaia che potrete contattare al numero 095-434534. Per comunicazioni urgenti potete inviare un sms al numero 3290517411. cordiali saluti Dott. Gianrico Fichera ITESYS srl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad recorded audio quality (upgrade).
Ken D'Ambrosio wrote: Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old Make sure you compile with the 'Don't optimize' flag if you're using gcc 4.2.2 Doug make sure you compile with 'Don't optimize' if you're using gcc 4.2.2 -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad recorded audio quality (upgrade).
On Sun, Aug 03, 2008 at 08:10:54PM -0400, Doug Lytle wrote: Ken D'Ambrosio wrote: Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old Make sure you compile with the 'Don't optimize' flag if you're using gcc 4.2.2 Doug make sure you compile with 'Don't optimize' if you're using gcc 4.2.2 Actually: set the optimizations to -O2 (what exactly is -O6? Where exactly is it defined) in the main Makefile or use the system version of libgsm and make sure you tell that to configure (--with-gsm=system or something similar, IIRC). On Debian this means installing libgsm-dev . IIRC this should not be required in latest SVN, but I'm not sure. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I issue a Flash to Zap (PSTN) from SIP?
Because of callee caller: inflash = *4,caller,Flash,() outflash = *3,callee,Flash,() On Sun, Aug 3, 2008 at 1:18 PM, Jim Duda [EMAIL PROTECTED] wrote: Cool, thanks for the tip. Why do you need to separate incoming and outgoing? Jim C F wrote: This is what I do: /etc/asterisk/features.conf [applicationmap] inflash = *4,caller,Flash,() outflash = *3,callee,Flash,() in /etc/asterisk/extensions.conf before accepting a call: exten = s,n,Set(DYNAMIC_FEATURES=inflash) on an outgoing call: exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash) in incoming calls the user has to press *4 on outgoing calls the user has to press *3 On Sat, Aug 2, 2008 at 4:59 PM, Jim Duda [EMAIL PROTECTED] wrote: I've seen a few posts on this issue, however, no definitive answer. My PSTN is connected to Zap/4. I have simple Call Waiting service on the PSTN line. All the other phones are SIP clients. When I'm on an Zap/SIP connection and another call comes in, I can hear the Call Waiting Tone on the SIP line. How can I issue a Flash/Hook to the Zap line in order to accept the other call? Also, is there any means to get Caller ID for the other call? I've seen posts that I can use *0 or *3 to send the Flash/Hook, however, that doesn't work for me. I realize there is a Flash( ) Dialplan function. How can I use this Function in the Dialplan with a call which is currently in progress? Any advice is most appreciated. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FC2 and Zaptel
Hi, I am using an older Fedora - FC2 and trying to install zaptel.(for X100P card I have - FXO with one line port and one Phone port) Fist I tried installin from RPM...as given here (also tried installing Zapata) http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora (FC2 is at the end) But looks like zaptel executable was not there...same for zapata... Then I started on downloading the source, I successfully completed MAKE for zaptel...but make install has following error...Full o/p follows: = [EMAIL PROTECTED] zaptel]# make install make[1]: Entering directory `/usr/src/zaptel' make -C /lib/modules/2.6.10-1.771_FC2/build ARCH=i386 SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/lib/modules/2.6.10-1.771_FC2/build' Building modules, stage 2. MODPOST *** Warning: class_device_destroy [/usr/src/zaptel/kernel/zaptel.ko] undefined! make[2]: Leaving directory `/lib/modules/2.6.10-1.771_FC2/build' make[2]: Entering directory `/usr/src/zaptel/kernel/xpp/utils' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/usr/src/zaptel/kernel/xpp/utils' make[1]: Leaving directory `/usr/src/zaptel' install -d /etc/udev/rules.d build_tools/genudevrules /etc/udev/rules.d/zaptel.rules build_tools/uninstall-modules dahdi 2.6.10-1.771_FC2 make -C /lib/modules/2.6.10-1.771_FC2/build ARCH=i386 SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install make[1]: Entering directory `/lib/modules/2.6.10-1.771_FC2/build' INSTALL /usr/src/zaptel/kernel/pciradio.ko INSTALL /usr/src/zaptel/kernel/tor2.ko INSTALL /usr/src/zaptel/kernel/torisa.ko INSTALL /usr/src/zaptel/kernel/wcfxo.ko INSTALL /usr/src/zaptel/kernel/wct1xxp.ko INSTALL /usr/src/zaptel/kernel/wct4xxp/wct4xxp.ko INSTALL /usr/src/zaptel/kernel/wctc4xxp/wctc4xxp.ko INSTALL /usr/src/zaptel/kernel/wctdm.ko INSTALL /usr/src/zaptel/kernel/wctdm24xxp/wctdm24xxp.ko INSTALL /usr/src/zaptel/kernel/wcte11xp.ko INSTALL /usr/src/zaptel/kernel/wcte12xp/wcte12xp.ko INSTALL /usr/src/zaptel/kernel/wcusb.ko INSTALL /usr/src/zaptel/kernel/xpp/xpd_fxo.ko INSTALL /usr/src/zaptel/kernel/xpp/xpd_fxs.ko INSTALL /usr/src/zaptel/kernel/xpp/xpd_pri.ko INSTALL /usr/src/zaptel/kernel/xpp/xpp.ko INSTALL /usr/src/zaptel/kernel/xpp/xpp_usb.ko INSTALL /usr/src/zaptel/kernel/zaptel.ko INSTALL /usr/src/zaptel/kernel/ztd-eth.ko INSTALL /usr/src/zaptel/kernel/ztd-loc.ko INSTALL /usr/src/zaptel/kernel/ztdummy.ko INSTALL /usr/src/zaptel/kernel/ztdynamic.ko INSTALL /usr/src/zaptel/kernel/zttranscode.ko make[1]: Leaving directory `/lib/modules/2.6.10-1.771_FC2/build' [ `id -u` = 0 ] /sbin/depmod -a 2.6.10-1.771_FC2 || : make[1]: Entering directory `/usr/src/zaptel/kernel/xpp/utils' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/zaptel/kernel/xpp/utils' make[1]: Entering directory `/usr/src/zaptel/kernel/xpp/utils' install -d /usr/sbin install genzaptelconf fpga_load zt_registration xpp_sync lszaptel xpp_blink zapconf zaptel_hardware /usr/sbin/ install -d /usr/share/zaptel install -m 644 ../firmwares/FPGA_1141.hex ../firmwares/FPGA_1151.hex ../firmwares/FPGA_FXS.hex ../firmwares/USB_FW.hex init_fxo_modes /usr/share/zaptel/ install ../init_card_1_30 ../init_card_2_30 ../init_card_3_30 ../init_card_4_30 xpp_fxloader /usr/share/zaptel/ install -d /usr/share/man/man8 install -m 644 genzaptelconf.8 fpga_load.8 zt_registration.8 xpp_sync.8 lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8 /usr/share/man/man8/ install -d /etc/hotplug/usb install -m 644 xpp_fxloader.usermap /etc/hotplug/usb/ # for backward compatibility and for hotplug users: ln -sf /usr/share/zaptel/xpp_fxloader /etc/hotplug/usb/ install -d /etc/udev/rules.d install -m 644 xpp.rules /etc/udev/rules.d/ install -d /usr/lib/perl5/site_perl/5.8.3 for i in Zaptel Zaptel/Xpp Zaptel/Config Zaptel/Hardware; \ do \ install -d /usr/lib/perl5/site_perl/5.8.3/$i; \ done for i in Zaptel.pm Zaptel/Chans.pm Zaptel/Hardware.pm Zaptel/Span.pm Zaptel/Utils.pm Zaptel/Xpp.pm Zaptel/Xpp/Line.pm Zaptel/Xpp/Xbus.pm Zaptel/Xpp/Xpd.pm Zaptel/Config/Defaults.pm Zaptel/Hardware/PCI.pm Zaptel/Hardware/USB.pm; \ do \ install -m 644 zconf/$i /usr/lib/perl5/site_perl/5.8.3/$i; \ done make[1]: Leaving directory `/usr/src/zaptel/kernel/xpp/utils' install -d /sbin install fxotune ztcfg ztmonitor ztspeed zttest ztscan zttool /sbin/ install -d /usr/share/man/man8 install -m 644 doc/fxotune.8 doc/ztcfg.8 doc/ztmonitor.8
Re: [asterisk-users] Asterisk Queues problem- URGENT
Hi, Can anyone help me on this. I am really stuck.again defining the problem briefly.: 1. Second New card TDM240P added to machine. 2. Only FXO modules i.e 24 FXO. 3. Asterisk detected all the ports successfully and when I run module reload chan_zap.so it list allthe FXO ports correctly. 4. when I can on any of the newly added lines there is a clear ring on the orginators phone while no activity detetcted by asterisk. It just keep quiet. It looks like call is not being detected by the card to my asterisk. 5. 4 port FXO card which was previously installed is functioning properly only this new added card is causing problem. 6. I have 12 new lines and only one of the lines is generating below mentioned logs in asterisk: == Starting post polarity CID detection on channel 18 -- Starting simple switch on 'Zap/18-1' [Aug 4 11:09:29] NOTICE[12255]: chan_zap.c:6169 ss_thread: Got event 17 (Polarity Reversal)... [Aug 4 11:09:31] WARNING[12255]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/18-1' == Starting post polarity CID detection on channel 17 -- Starting simple switch on 'Zap/17-1' [Aug 4 11:09:35] ERROR[12256]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-1) [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6202 ss_thread: CallerID feed failed: Success [Aug 4 11:09:35] WARNING[12256]: chan_zap.c:6215 ss_thread: CallerID returned with error on channel 'Zap/17-1' [Aug 4 11:09:37] WARNING[12256]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/17-1' Can anyone decipher this code??? What is happening?? Please give me some cluess to work on. In my Zapata.conf I have following two lines related to above logs: Cidsignalling= v23 Cidstart = polarity Please help./ Syed nasr (MONDAY 04/08/2008) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Syed Nasruddin Sent: Friday, August 01, 2008 8:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Thanks, Yes that was the problem I have added joinempty=yes. It is now working,. Right now another critical problem has come up which I have mentioned in my previous email. I am copying the problem here again: was initially running only with one TDM800P card having 4FXO and 4 FXS port then I later added another 24 port FXO card. So now in total I have now 32 FXO ports for in coming calls. Card was successfully integerated and all the ports were detected by asterisk. Just few minutes back the POT lines were also ready and now I am getting additional errors which I am pasting here. starting simple switch on 'Zap/17-1'[Aug 1 19:00:26] ERROR[3416]: callerid.c:564 callerid_feed: fsk_s erie made mylen 0 (-1)[Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6202 ss_thread: Caller ID feed failed: Success [Aug 1 19:00:26] WARNING[3416]: chan_zap.c:6215 ss_thread: Caller ID returned with error on channel 'Zap/17-1' [Aug 1 19:00:28] WARNING[3416]: chan_zap.c:6232 ss_thread: CID ti med out waiting for ring. Exiting simple switch-- Hungup 'Zap/17-1' -- Saved useragent X-Lite release 1002tx stamp 29712 for pee r 1001[Aug 1 19:18:29] NOTICE[3162]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 17 == Starting post polarity CID detection on channel 17-- Starting simple switch on 'Zap/17-1' [Aug 1 19:18:29] NOTICE[3582]: chan_zap.c:6169 ss_thread: Got event 4 (Alarm)... [Aug 1 19:18:31] WARNING[3582]: chan_zap.c:6232 ss_thread: CID timed out waiting for ring. Exiting simple switch Hungup 'Zap/17-1' Please help on this urgent. I cant upgrade right now since I am not confident abt upgrade procedure and any other problems occuring after that. This is my only production machine. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson Sent: Friday, August 01, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Queues problem Syed Nasruddin wrote: Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: _app_queue.c:3939 queue_exec: unable to join queue myqueue_ In extension file: Queue(myqueue|t|||120) And my agents are joining in following manner: Exten = 1001,1,AgentLogin(SIP/1001) Exten = 1000,1,AgentLogin(SIP/1000) One more thing my asterisk successfully captures the call , it plays music on hold but when it starts to push the call in queue it gives out this error. Any one help me out. It's a production machine. Thanks Syed nasr
