Re: [asterisk-users] USA Did required
In the USA ordering BRI service is discouraged by the telcos and is very uncommon. In Verizon NE CLECs are not even permitted to order BRIs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: Friday, September 30, 2011 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] USA Did required Go to ATT and ask for a BRI ISDN Line. Tell them that you want a system access with a number block. that would be typically 54448[0-9] where your extensions are from 0-9 I don't know what protocol the americans are using, as I know the americans made for sure their own thing which is known as the north american BRI protocoll. If they have replaced it with the euro isdn protocol, I don't know. Figure out in advance if the supplier, in this case Digium, Sangoma or Beronet support the north american protocol. in Germany we use the EUROISDN protocol. Here in Germany it is usual to have such lines, we don't use VoIP with DID. logicly it could be possible, but I never saw a provider. By the ISDN is more secure. If ISP connections fall out, you are through your telephone lines still reachable. Information: For BRI Isdn you might need a NT Unit where you would connect the isdn cable to your board. Tamer Am 01.10.2011 02:29, schrieb Tarek Sawah: Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- -- Date: Sat, 1 Oct 2011 00:51:59 +0530 From: amit.magn...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] USA Did required Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterik-users] Installing PRI card
On Sat, Oct 01, 2011 at 05:15:32AM +0530, NaJIm wrote: *wcte12xp: Setting up global serial parameters for T1* *wcte12xp: Found a Wildcard TE122* *wcte12xp: Span configured for ESF/B8ZS* *wcte12xp: Setting yellow alarm* Does the highlighted part mean that the card is setup in T1 mode. ?? ( I do not have physical access to the server. But my support in the remote office says that card is in E1 mode itself.. ) Sit back, take a breath and relax, stop trying things at random. First you have to figure out what signalling has to used on your PRI. What country are you in? Who is the provider? Find out the switchtype and siganalling used in zapata.conf and the spantype for zaptel.conf. Also the B channels must match and offcourse the D channel must be set correctly. At the moment you are using typical T1 settings (24 channels, ESF/B8ZS) but also say the thing is an E1 (30 channels (wheter or not you actually get the max number of voicechannel), CCS,HDB3(,CRC4)). T1 is the default on Digium cards, so you might had to set the jumper differently, but since you don't have physical access you need to use the correct module option (t1e1override on dahdi, might differ on zaptel (see modinfo)). -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Converting dahdi_monitor unit to dbm0
Hello, I need to convert the dahdi_monitor output to dBm0, so I can measure Echo Return Loss in dB. I've read a formula that calculate S(k) in ITU-T G.168 recommendation, where S(k) is the signal level in dBm0. Can I use this formula to convert it? If yes, what value should I use to the number of samples if I want to convert a single output from dahdi_monitor? -- Thanks in advance, Gustavo Santos. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA Did required
I say only, that it is MONEYMAKING ISDN is the best thing ever. Using the internet completly through VoIP is wasting ressources and if the ISP connection falls for a time, the line communication fall either. I think if a state permits ordering isdn bri channels, then there is only a thought behind to make more money for each lines that is being ordered or to sell for high prices ISDN PRI channels, which is in my view nothing else as a unfair isolation business policy. Sorry, but SMB need to work efficient to grow with telephony without falling in high expenses. I spoke with a friend of mine in L.A. and in the US professional telephony services are much more expensive as in Germany. I still don't know why. Tamer Am 01.10.2011 18:59, schrieb Eric Wieling: In the USA ordering BRI service is discouraged by the telcos and is very uncommon. In Verizon NE CLECs are not even permitted to order BRIs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: Friday, September 30, 2011 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] USA Did required Go to ATT and ask for a BRI ISDN Line. Tell them that you want a system access with a number block. that would be typically 54448[0-9] where your extensions are from 0-9 I don't know what protocol the americans are using, as I know the americans made for sure their own thing which is known as the north american BRI protocoll. If they have replaced it with the euro isdn protocol, I don't know. Figure out in advance if the supplier, in this case Digium, Sangoma or Beronet support the north american protocol. in Germany we use the EUROISDN protocol. Here in Germany it is usual to have such lines, we don't use VoIP with DID. logicly it could be possible, but I never saw a provider. By the ISDN is more secure. If ISP connections fall out, you are through your telephone lines still reachable. Information: For BRI Isdn you might need a NT Unit where you would connect the isdn cable to your board. Tamer Am 01.10.2011 02:29, schrieb Tarek Sawah: Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- -- Date: Sat, 1 Oct 2011 00:51:59 +0530 From: amit.magn...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] USA Did required Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Make asterisk cluster appear and operate as a single server?
Hi, I'm trying to plan a system of clustered asterisk machines where a number of SIP trunks will be hosted on the platform. Each trunk will be hosted for a specific customer who owns it and therefore payment is handled directly between the customers and their trunk-providers, each trunk will have about 50-200 simultaneous calls. No SIP phones will be directly connected to the platform, my thought is that the asterisk machines should only receive incoming and make outgoing calls through the trunks, and then connect the calls with each other. To make this scalable and have the option of running an infinite number of sip-trunks, I need a good way to load-balance my asterisk servers and implement failover support and also be able to add / replace the machines in the cluster in a safe and reliable way. I'm have some experience building single asterisk solutions but I have never worked with load balancing of multiple asterisk machines. Is it possible to configure all trunks on a single asterisk setup which is then reflected over a cluster of asterisk machines? If I have a cluster of machines, I guess I need some kind of front-end application / system? I will then also need to be able to connect calls between the machines, the calls to be connected with each other will always be incoming and outgoing on the same trunk. In other words, I want to create a large cluster of asterisk machines to appear and operate as a single asterisk server. I've looked at projects like OpenSIP but it feels like this is not really what I need? I really appreciate if someone can help me get on the correct path here, I need all the feedback I can get. Thanks in advance! Best regards Tobias -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?
If one server is supposed to carry the full load of the other during failure, then you have to size each server to handle 100% load - so load balancing is pointless. Checkout haast at www.generationd.comhttp://www.generationd.com and read the docs on how it does failover...certainly good for ideas. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Tobias Steen [tobias.st...@s2.se] Sent: Saturday, October 01, 2011 6:30 PM To: Asterisk Users List Subject: [asterisk-users] Make asterisk cluster appear and operate as a single server? Hi, I'm trying to plan a system of clustered asterisk machines where a number of SIP trunks will be hosted on the platform. Each trunk will be hosted for a specific customer who owns it and therefore payment is handled directly between the customers and their trunk-providers, each trunk will have about 50-200 simultaneous calls. No SIP phones will be directly connected to the platform, my thought is that the asterisk machines should only receive incoming and make outgoing calls through the trunks, and then connect the calls with each other. To make this scalable and have the option of running an infinite number of sip-trunks, I need a good way to load-balance my asterisk servers and implement failover support and also be able to add / replace the machines in the cluster in a safe and reliable way. I'm have some experience building single asterisk solutions but I have never worked with load balancing of multiple asterisk machines. Is it possible to configure all trunks on a single asterisk setup which is then reflected over a cluster of asterisk machines? If I have a cluster of machines, I guess I need some kind of front-end application / system? I will then also need to be able to connect calls between the machines, the calls to be connected with each other will always be incoming and outgoing on the same trunk. In other words, I want to create a large cluster of asterisk machines to appear and operate as a single asterisk server. I've looked at projects like OpenSIP but it feels like this is not really what I need? I really appreciate if someone can help me get on the correct path here, I need all the feedback I can get. Thanks in advance! Best regards Tobias -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
