Re: [asterisk-users] DUDE!!!!! was RE:Dialplan Visualization(Extensions.conf or Dialplan Show)
So this is what it has all come down to? Spamming the mailing lists... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, April 18, 2008 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUDE! was RE:Dialplan Visualization(Extensions.conf or Dialplan Show) Lol - you really do hate anyone doing anything commercial with asterisk huh :) Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Friday, 18 April 2008 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUDE! was RE:Dialplan Visualization(Extensions.conf or Dialplan Show) My post was made b/c John Signorello has done this before and I thought that a friendly reminder of the proper places to post his 'offers' should be posted. This is the one that came to mind when I composed the email reply: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg202014.ht ml -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Friday, April 18, 2008 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUDE! was RE: Dialplan Visualization(Extensions.conf or Dialplan Show) Brent Davidson wrote: John Signorello wrote: excuse me... But did you not just post [asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New Digium Boards Cheap X305 $199 Did you not provide a link to a COMMERICAL entity? Wasn't your a post a unsolicited post, that is, not in response to a question??? There seems to be two standards here. The fact that you do not work for them is immaterial. If your argument is no commercial reference at all, then how do you explain your post? I felt like the cogoblue stuff is out of place and off topic, so I agree that it should not be arbitrarily posted as a [EMAIL PROTECTED] solution to any question that it might seem to somehow solve. However, I do not take exception to the IBM server post, even though strict adherence to the rules would probably make it illegal as well. I think most people agree (or a least don't mind too much) if a commercial product is offered as a possible solution to an OP's query, assuming it is in fact, within context. I'll leave it to others on the list to decide if John Signorello's post was appropriate or not, given the context of the OP's original query, but if someone posts a query directly related to a product I have to offer, I fully intend to let them know about it in as least intrusive manner as I can. Assuming a person's product is directly within context, not offering it as possible solution could be a disservice to the OP and list in general. Just a thought... -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- 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 ___ -- 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] Cisco power injector with GXP2000 phones
The electricity is carried on different pins in a cisco poe injector. Just because they both support the same standard doesn't mean they were implemented the same. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Thursday, December 06, 2007 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco power injector with GXP2000 phones I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho. ___ --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] Cisco power injector with GXP2000 phones
That phone supports POE. However, there is some talk of a known issue that they tend to crash after 1-2 hours on POE. Grandstream phones are quite horrible products. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Thursday, December 06, 2007 10:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco power injector with GXP2000 phones Quoting Salvatore Giudice [EMAIL PROTECTED]: The electricity is carried on different pins in a cisco poe injector. Just because they both support the same standard doesn't mean they were implemented the same. I didn't even know the gxp2000's could handle poe - anyone care to share what voltage/current they expect on what pins ? even if I could just move the existing powersupplies back to the punchdown panel that would unclutter desks and make centralizing ups power that much simpler. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Thursday, December 06, 2007 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco power injector with GXP2000 phones I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Cisco power injector with GXP2000 phones
I forgot to answer your other question. I don't have the pin outs, but the way cisco proprietizes the POE injector involves reversing the polarity carried on the cable. Literally, two pins are switched going into and coming out of the POE injetcor. You could use a cisco poe injector with 2 custom Ethernet cables that crossover those two pins to correct the polarity difference. You should be able to figure out which pins cisco uses with a cheap amp meter form radioshack. There are also a lot of generic poe adapters out there that will work fine with you GXP2000/asterisk setup. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Thursday, December 06, 2007 10:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco power injector with GXP2000 phones Quoting Salvatore Giudice [EMAIL PROTECTED]: The electricity is carried on different pins in a cisco poe injector. Just because they both support the same standard doesn't mean they were implemented the same. I didn't even know the gxp2000's could handle poe - anyone care to share what voltage/current they expect on what pins ? even if I could just move the existing powersupplies back to the punchdown panel that would unclutter desks and make centralizing ups power that much simpler. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Thursday, December 06, 2007 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco power injector with GXP2000 phones I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] Cisco 7960 to 2 SIP servers?
It can be attached to 6 if I remember correctly. However, each is a separate line. Cisco will not perform a seamless connection to multiple servers for a single line as some sort of fail over system. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Thursday, December 06, 2007 11:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 to 2 SIP servers? Yes it's work for me... (with olds 7940 phones...) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Shawn Laemmrich Envoyé : mercredi 5 décembre 2007 23:43 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Cisco 7960 to 2 SIP servers? Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2 different SIP servers @ the same time? I currently have an asterisk box @ home with several sip extensions and a Nortel C15k phoneswitch at work (not the pbx, the full phone switch). I can connect from the SIP phone to the Nortel phone switch, but cannot make asterisk talk to it at all (if anyone has any ideas on this one, I'd be hugely grateful). So I thought if I could have the cisco ip phone on my desk talk to both servers (like a line1 is my home asterisk server, line 2 is the nortel switch) I'd be all set. Does anyone know if this is possible, and if so how to do it? Thanks in advance Shawn ___ --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] Outgoing PSTN calls , unusable voice quality
Well do you have a packet filter between the asterisk box and your phone? Is the phone or the asterisk behind a NAT? Do you have an asymmetric route for traffic in your network? Does the media take the same path inbound and outbound between the asterisk and the phone? If you take a packet capture of the voip segment of both an inbound and outbound call, how do they differ? Are those calls using different codecs? If you want to rule out the analog card, construct a loop. When an inbound call comes in, send it back out to the PSTN. Then create the call and get a MOS score if you can. You should also consider terminating an inbound and an outbound call against the echo application in asterisk and see if you can further isolate the problem to either the PSTN channel or the voip channel. You really need to try to isolate the problem to a particular call segment. Once you do that, make sure your test is repeatable. I would be very hesitant to declare a hardware issue with the Sangoma card. Unidirectional audio problems are usually caused by a Voip leg since media is handled separately during that segment of the call. I would be extremely surprised if this was a hardware issue with a sangoma card. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Sunday, December 02, 2007 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality Dear Salvatore and Joanna, Thank you much for both your detailed explanations. I will surely check my firewall configuration and logs to make sure the VoIP traffic is passing correctly. However, I'm a bit confused as the problems that I'm experiencing are with calls made via the Sangoma analog card. So the voice goes from the SIP phone through asterisk through the sangoma card then directly into the PSTN and vice versa. There are no firewalls in the way. Furthermore incoming calls are OK, the problem is only with outgoing calls when I can hear the other party well but they barely understand me. Is there any major difference in the way that Incoming/Outgoing calls are processed in the above scenario? Any way that I could trace those processes for faults? Thank you again. Regards, Veselin Salvatore Giudice wrote: When you take your packet capture, you'll need to look at the sip messages with SDP attached to get the ip's and ports used for both media streams. Make sure that the ips are correct and that the port used can traverse between those ip's without being blocked by a packet filter or firewall. A lot of times, administrators will set a range of UDP ports that are allowed to pass their packet filter for media and your pbx or phones may be using a different range. This can cause audio loss. You'll need to eliminate that possibility. Sometimes checking your firewall/packet filters for blocks may also prove helpful in identifying problems. You should be aware that the logs from certain firewall products may not be comprehensive. For example, in the past I have seen packets dropped going through netscreens because of invalid headers and no entries appeared in the logs. If you ultimately believe a firewall may be blocking your traffic make sure you setup a capture port or a span on each side of the device and verify the traffic going to and leaving from the firewall using ethereal on a laptop or maybe a Nixon box if you are in a large distributed environment. Never trust a potentially broken device to report accurate information about it's function. TDM = Time Division Multiplexing TDM describes how channels are separated on T1's, etc. It's common to refer to those types of connections as TDM. http://en.wikipedia.org/wiki/Time-division_multiplexing -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all
Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
When you take your packet capture, you'll need to look at the sip messages with SDP attached to get the ip's and ports used for both media streams. Make sure that the ips are correct and that the port used can traverse between those ip's without being blocked by a packet filter or firewall. A lot of times, administrators will set a range of UDP ports that are allowed to pass their packet filter for media and your pbx or phones may be using a different range. This can cause audio loss. You'll need to eliminate that possibility. Sometimes checking your firewall/packet filters for blocks may also prove helpful in identifying problems. You should be aware that the logs from certain firewall products may not be comprehensive. For example, in the past I have seen packets dropped going through netscreens because of invalid headers and no entries appeared in the logs. If you ultimately believe a firewall may be blocking your traffic make sure you setup a capture port or a span on each side of the device and verify the traffic going to and leaving from the firewall using ethereal on a laptop or maybe a Nixon box if you are in a large distributed environment. Never trust a potentially broken device to report accurate information about it's function. TDM = Time Division Multiplexing TDM describes how channels are separated on T1's, etc. It's common to refer to those types of connections as TDM. http://en.wikipedia.org/wiki/Time-division_multiplexing -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --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 ___ --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] Off-Topic: Avaya
Avaya makes 52% of it's revenue from professional services. In enterprises, you generally have 3 budgets: Captial, expense, professional services Avaya figured out that they could make more money tapping into professional services portion of the budget with charge by the hour union consultants than by selling equipment. Avaya is also the most pervasive vendor in the space when it come to calling dev products GA, so they can get their customers to pay them to beta test. Avaya's newest ploy is to get customers hooked on their systems and after 6 - 12 months of shear hell supporting the products, they kindly offer to outsource your voice infrastructure support using a system called SIG. SIG requires you to place a collector box on your network with an IPSEC VPN nailed up to Avaya corporate. This gives them full unchecked access to your network. Exciting huh? Introducing Avaya into a corporate network is about as smart as introducing syphalis into a high school. Sure, it was all fun and games at first, but eventually it catches up to you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Molina Sent: Saturday, December 01, 2007 1:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Off-Topic: Avaya Salvatore Giudice wrote: They are cheap. You only have to pay for the box and the maintenance percentage. That is indeed the Avaya way. First you buy it, then you rent it. Stop paying their maintenance fees and their dial into your PBX and cripple the OS by removing customer maintenance command permissions. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. Bravo. A well-deserved lambasting of this awful vendor. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --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] Outgoing PSTN calls , unusable voice quality
Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --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] Off-Topic: Avaya
If you desire SIP in Avaya, you have to add a SES (SIP Enablement Server) to your Avaya setup. They are cheap. You only have to pay for the box and the maintenance percentage. You don't need to buy user ports or any of that garbage as long as you setup your extensions using Optum, which is a free Avaya feature. The SES maintains a registry and a dial plan. SIP phones attached to SES send media directly to medpros and the SES does a protocol conversion between SIP and H.323 to bridge a connection between the SIP phone and the CLAN cards. The voicemail issue you describe with the MWI is because Avaya's systems use qsig trunks to connect to voicemail servers. Asterisk is not connected int hat manner, so of course you won't be able to support Avaya MWI's. However, you can deposit a script on your asterisk that would send the standard notifies to the Avaya phones to manipulate the MWI's directly. However, you will need to statically address the phones and keep track of them because you cannot poll an SES server for their SIP URI's. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Friday, November 30, 2007 9:54 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Off-Topic: Avaya This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --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] zaptel huge irq problem
Try switching to a Sangoma card. You wont have anymore IRQ issues once you abandon Digium hardware. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] zaptel huge irq problem Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? François. Michael L. Young wrote: François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 10:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel huge irq problem Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 _ La información contenida en este mensaje y en sus archivos adjuntos es CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda expresamente prohibida la utilización de la misma por cualquier persona distinta de los destinatarios de esta
RE: [asterisk-users] Asterisk x legacy pabx
Ive done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you cant use Asterisk in the same way you can use a modular messaging setup. Asterisk will only work if you actually terminate the employees phone on the asterisk box and that would be kind of pointless because businesses only want Avaya because eof the extra feature they offer. You would of course lose most of them if you were just using Avaya to manage the tie lines to an Asterisk box. On the brighter side, I would bet your licensing would be a hell of a lot cheaper I worked with Avaya for 3 years prototyping solutions involving their CCS/SES product line. Their stuff does not play well with other equipment. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk x legacy pabx
Its basically the same problem. Asterisk is not a standalone voicemail server. It would have to support Qsig. Asterisk doea not exactly have expansive Qsig support. I believe there are several bounties out for Qsig. Without Qsig, you would have to use parallel forking and ring the users avaya or siemans extension and also the same extension on an Asterisk box. Youd have to manage a dummy number for every mailbox configure dont he asterisk box. Also, I dont know for siemans, but Avaya doesnt support parallel forking, so you would have to either configure both the employee and asterisk as an optum extension or buy an x-mobility/extension-to-celluar license to accomplish either. I think x-mobility is $300 list per phone. Its horribly expensive. Among Nortel, Avaya, Mitel voicemail systems Mitel is by far the best product of these 3. Avaya message networking/modular messaging is basically a beta. Nobody should consider that GA. Its horrible. Nortel requires too much professional service money to get up and running. Nortel seems to think they can charge 4 times more for everything because it says Nortel. I havent figured that one out yet. Mitel was my preferred vendor voicemail product since it is reasonably priced and their support organization is actually attentive. Check out Mitel 10. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk x legacy pabx Hi Salvatore, thanks for reply. And if pabx legacy was Siemens model HiPath 3750, could use asterisk as serving of voicemail and other applications? Best Regards Josué 2007/5/5, Salvatore Giudice [EMAIL PROTECTED]: I've done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you can't use Asterisk in the same way you can use a modular messaging setup. Asterisk will only work if you actually terminate the employee's phone on the asterisk box and that would be kind of pointless because businesses only want Avaya because eof the extra feature they offer. You would of course lose most of them if you were just using Avaya to manage the tie lines to an Asterisk box. On the brighter side, I would bet your licensing would be a hell of a lot cheaper I worked with Avaya for 3 years prototyping solutions involving their CCS/SES product line. Their stuff does not play well with other equipment. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com http://voipsecuritytraining.com/ 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti Sent: Saturday, May 05, 2007 1:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk x legacy pabx Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josué ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk telemarketer torture sound files
Just forward them to 1-800-big-dick or some other 800 toll free phone sex line. They can't tell they've been forwarded. They'll figure it out eventually. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Jacob Muller Sent: Saturday, May 05, 2007 1:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk telemarketer torture sound files Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
My money is on compulsory drug rehab or simply being held for 45 days of observation after being caught sexually abusing a pony. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: Saturday, May 05, 2007 4:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? At the very least, he's abusing his customers. Substances? I hadn't thought of that. On 4/30/07, Salvatore Giudice mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Codec Translation Table
It's the magical Celeron chip. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Sent: Friday, May 04, 2007 3:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Codec Translation Table Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 gsm- -223 21 9 - - 253 - ulaw- 6 -13 21 9 - - 253 - ilbc-106676513 - --7- g726- 73313210 - -26-- Second server is Dual Xeon 2Gh 1G RAM show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g72 gsm- -446 4328 - - 255- ulaw- 7- 14 2126 - - 233- ilbc- 9446 4328 - - - 5- g726-7 221 2126 - - 23 -- Here is the fun part, box1 is faster in converting ulaw to gsm! Is this table accurate? Does it mean asterisk is not handeling multiple cpus very good? both boxes running asterisk 1.4.4 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 0 duration but non-zero billsec in mysql cdr
That's a feature to generate more revenue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Thursday, May 03, 2007 4:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 0 duration but non-zero billsec in mysql cdr I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they had same record with duration 0 and higher bill seconds . Happen with both asterisk 1.2.17 as well as 1.2.18 All sip to iax/sip calls . Destination numbers were valid. Dialplan maintained with freepbx . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 0 duration but non-zero billsec in mysql cdr
Roflol. How about a script that makes calls for people after 15 min of inactivity... Streamline the whole process. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Thursday, May 03, 2007 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 0 duration but non-zero billsec in mysql cdr Someone in -biz list pointed out that this could be a freepbx problem so i think i will go check there . @ Salvatore Giudice: how can i intentionally do it ? Damn i need a app that can make sure customer phone doesnt hangup for the time i specify .. even if customer breaks his phone . lol ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] you have been kicked my this conference
Replace it with a pause sound byte. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, May 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] you have been kicked my this conference How do I stop the you have been kicked by this conference message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX Trunk
Good luck. Try these. http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/IAX+encryption http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, May 03, 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linseed
Interesting play for maybe a smartphone, pda , or internet enabled microwave oven . It's a little expensive for that price. I just picked up a new half-ebx Advantech pcm-9381 with 1GB DDR 333 and a Pentium M 745 1.8Ghz cpu for $250. That's smaller than the evaluation board they picture in that article. Plus, it features 2x18 bit LVDS and pc/104. On the downside it will likely generate more heat. I need to find a 2GB compact flash card and a 10.4 touch screen lcd so I can start building a phone prototype. I just wish I could find a chassis and a small power supply for it. MBOX was making them for Advantech gear, but I can't seem to find one for this board model. =( I think these kind of soft chips may end up being popular in PDA's, but only if manufacturers can get them for less than $15. The majority of $400 - 500 cell phones on the market cost less than $30 to manufacture with the lcd being the most expensive component. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, May 03, 2007 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Linseed http://www.linuxdevices.com/news/NS3013985136.html Ok so who's going to be the first to install Asterisk on it? Regards, Dean Collins Cognation Pty Ltd mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation Call Button image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX Trunk
Yes of course. If you want to limit it, I think you have to set 'incominglimit' and/or 'outgoinglimit'. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, May 03, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? Thanks. Steve Kennedy wrote: On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE
I don't think you can do that. You can easily issue a 302 with something like SER or OpenSER. I believe the only thing Asterisk can do is receive a call on the initial URI and open a channel to the destination and connect them. Media could pass directly between those two points but your Asterisk box would still have to participate in the signaling. Think of Asterisk as a B2BUA instead of a SIP call router/response system. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Thursday, May 03, 2007 6:18 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE From: CSB [EMAIL PROTECTED] Date: Thu, 3 May 2007 21:51:02 +1200 I want to get Asterisk to redirect an incoming SIP INVITE to another SIP URI. I was looking at the Transfer application but it seems to You may want to elaborate the requirement. How is the incoming INVITE initiated? Is the originator a user in your system? Does the other URI represent a peer? etc. Yuan Liu be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). Is there an alternative way to do this on Asterisk 1.2.18? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
That phone sounds like a real pain in the ass. Besides, it looks a little junky from the photos. I guess you really can't complain too much about a $150 light office SIP phone. BTW, how does the phone 'feel'? When you pick up the handset do you immediately get the feeling that it's a cheap phone? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang Sent: Thursday, May 03, 2007 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-Polycom HEPPP The polycom lets you do either attended or unattended transfers. If you want an unattended transfer, you press the 'blind' soft key. It's been a few months since I've looked at this, so a bit fuzzy on the details. Jason Adams wrote: Isn't that the function of an attended transfer? User3 hears User1 to see if they want to take the call or not. User1 should then hit the transfer key again to finalize the call. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jim Suber *Sent:* Thursday, May 03, 2007 12:54 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk-Polycom HEPPP PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 and Cisco Phones 7940
The features you mentioned will work fine, but you'll need to also have to maintain a tftp server to provision the phones and ensure a quick boot. In my experience, the only annoying thing about sip loads on Cisco phones is that they don't support sidecars for admin. http://www.voip-info.org/wiki-Setup+SiP+on+7940+-+7960 Make sure you remove any callmanager related info from your DHCP scope before you deploy Asterisk if they previously had a callmanager installed. When completing these types of conversions, you run the risk of the phones going to an unprovisioned state if they start trying to access a callmanager that has been removed from the network. It sucks to get called back to a job a few weeks later when the customer's phone gets whacked after it was unplugged and rebooted. Good luck -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Thursday, May 03, 2007 8:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4 and Cisco Phones 7940 I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- Erick Perez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to Execute System Command From DialPlan
Could it be permissions? What uid is your Asterisk running under and what are the perms for your sounds directory? sounds is this by default on my server since I built from source: drwxr-xr-x 13 root root 110592 Apr 14 01:13 sounds Try putting: /bin/mkdir -p /var/lib/asterisk/sounds/1234 /tmp/logfile 21 See if that generates a log for you at least. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of vcomp Sent: Thursday, May 03, 2007 8:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to Execute System Command From DialPlan Hello, I have scoured the mailing lists and forums to no avail. Does anyone have tips on how to use the system command within a dialplan (1.2.7.1). I am very familiar with dialplan scripting but new to the system command. I am attempting to create a directory. I put both of the lines below in my dialplan but neither executes, although they do not generate errors. The first line was added just for kicks to see if system is working properly. exten = s,n,System(/bin/pwd location.out) exten = s,n,System(/bin/mkdir -p /var/lib/asterisk/sounds/1234) Any assistance would be greatly appreciated. Thanks, Victor P.S. I received a suggestion to change System(/bin/pwd... to System(!/bin/pwd ... but it did not work, with or without a space. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Applet?
I have never 'managed' a hotel PBX. That is completely inaccurate. You have no idea what you are talking about. Yes, Estara was $2k dollars according to the client. If there were other provisions in their contract with Estara, I am not aware of them. They have made deals in the past that involved both cash and trade. That may very well have been the case for that specific deal. I wish you the best of luck in peddling your CTC product. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, May 02, 2007 1:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Applet? Yeh I know but it's just frustrating me, this is the third time that when someone has asked about web Click to Talk solutions Salvatore has piped up with Estara because he used to manage a hotel PABX where they used it. And that's all cool - it's a public list, but it's a $50,000 solution not $2,000 like he states. Anyway doesn't matter, I've been speaking with Pablo in Argentina backwards and forwards all night and after wasting time with Jiax and not getting it to work he's going to try out Corraleta we just need to work out some arrangements first. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, 2 May 2007 1:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Applet? Dean Collins wrote: Dude yes we know Estara is so cheap you said this 2 months ago, you said this last month and you are saying this today. Yet every customer that comes to us to buy a license says their quote was around $50,000 for the first year for Estara click to talk. I'm prepared to send a $100 bottle of wine to anyone who is able to show me a 1 year subscription to Estara's Active-X Click-to-Talk solution for $2,000. And yes Salvatore I'll be in Nevada in 3 weeks so will even personally deliver it if you are the one to provide the proof. You keep talking but you don't deliver. (BTW when you call Estara's sales team tomorrow tell them Dean from Mexuar sent you so they know where to send the Thank-You's for all the quotes to). Any replies take it to the biz list as this topic is dead. I hate to say this, Dean, but your approach is not helping sell your product. Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Capture Asterisk traffic
Sounds like you have an old libpcap. Try using this: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 'udp[2:2] = 5060 and udp[2:2] = 65534' This works on one of my machine that has a libpcap that doesn't support portrange. I guess you can't use macros to define the port range. So, you'll have to reference the header values directly. 0:2 is src port and 2:2 is dst port. Try that. It may work. Or you could try to upgrade libpcap. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Wednesday, May 02, 2007 4:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Capture Asterisk traffic I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 Thanks tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst portrange 5060-35000 tcpdump: unknown host 'portrange' tcpdump version 3.8 libpcap version 0.8.3 man tcpdump indicates that I should be able to use = syntax but it doesn't work as expected. Any further advice appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digital Phones
Yeah, they still sell the phones: http://products.nortel.com/go/product_content.jsp?segId=0catId=nullparId=0 prod_id=8593locale=en-US I looked around and I can't find that kind of multiplexer. You'd looks like you would need a small digital PBX since you need to be able to define stations and a dial plan. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, May 02, 2007 1:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital Phones Salvatore Giudice wrote: Nortel digital Meridian phones are like $400/each. At least that was the price of the phones at a hotel I did a job for recently. Still? (Is Nortel even making these phones anymore? I thought they spun off their telephone set division -- anybody heard of Aastra? ;) ) When you go to SIP, you may save on the capital costs for the phones, but other costs will increase. These are related to: 1.) increased support requirements for supporting VoIP 2.) additional licensing may be required from your vendor to support SIP or IP media 3.) increased costs associated with mitigating potential damages since your voice services are now subject to the same outages as your network 4.) increased training costs for staff to become proficient in VoIP 5.) increased costs associated with monitoring QoS 6.) increased costs associated with reconfiguring your network for VoIP and QoS - many times new switches may have to be purchased in addition to SBC's etc 7.) additional costs associated with rewiring physical space to accommodate additional Ethernet ports required for phones Yes, 7 times. Sometimes, if you already have digital - it may not be worth switching to SIP even if you save a ton on the handsets. Whenever we switch over a hotel to VoIP, we always run into these extra 'hidden' costs. If you want to do digital with asterisk, I think you'll need a T1/E1 multiplexer that supports digital phones. Is this anything like a channel bank, only for digital phones? Can you suggest any examples? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: How many users can be supported simultaneously?
CPU becomes more important if there is a lot of codec conversion. Memory becomes more important in supporting the overall volume of calls. Network resources are obviously limited by bandwidth. A 2.4ghz xeon with 1gb ram can easily handle 80+ calls is there is no codec conversion or the call terminates to a TDM card with an onboard DSP. If the box had to do codec conversion or emulate a DSP I would likely add a second CPU. In the past we looked at setting up high density gateways using 3 x Sangoma 8-port T1 cards and a dual/quad xeon 3.0ghz with 12gb of memory. Theoretically, it should work, but we never got around to testing it. Anyone setup a high density gateway and operated it under load? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Wednesday, May 02, 2007 12:20 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: How many users can be supported simultaneously? KM == Knud Müller [EMAIL PROTECTED] writes: KM Hi, there are some interesting figures on KM http://www.thrallingpenguin.com/articles/asterisk-solaris.htm. It's hard to take them as more than a lower bound on that particular hardware. No attempt is made at figuring out what actually limits throughput, and the cpu figures add up to more than 100%. Hopefully noone who cares about throughput uses a Celeron, anyway. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] allowing call every 15mins
Wow... Now that is customer service... I love it. It's the old maybe the
customer will stop calling if we stop answering the phone approach. I love
it.
Anyway, I think you could do it with an AGI call to a script that tracks
callerid and last call time. The script could basically decide whether to
pass them to a dial command or to a sound byte/hangup. For example the perl
AGI supports the following:
#for dialing
$AGI-exec('Dial', $option);
#for playing a sound byte
$AGI-stream_file('wedonnotwanttotalktoyou');
There are several AGI classes available for a variety of
scripting/programming languages. You would just need to mock up something
with callerid:time tracking and a simple check.
http://www.voip-info.org/wiki-Asterisk+AGI
Good luck, SG
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goke Aruna
Sent: Wednesday, May 02, 2007 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] allowing call every 15mins
Hello all,
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is 15, 30, 45, 60(in multiples of 15 minutes).
how can i achieve this and what application can i use to get this done.
I will be glad, if someone can give me a hint on this.
i have asterisk-1.12.1
zaptel-1.9.1
chan_ss7-0.8.4
Goksie
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RE: [asterisk-users] VPN between Asterisk server and phone client
SSL VPN's can be a bit sketchy when it comes to QoS. Usually IPSEC is
recommended for udp streaming media. However, people have shown some decent
success with SSL VPN's and VoIP. Free S/WAN is a good option if you want to
try IPSEC. It should be much more UDP friendly.
The following aren't VPN's. They are more like encrypted data pipes:
Zebeedee is also a fun option for encrypted, compressed tunnels suitable for
UDP. http://www.winton.org.uk/zebedee/ You can do some fun stuff when you
setup IAX on an internal interface with a Zebeedee listener. It's not for
the faint of heart though since setup can be a bit encumbering.
Some people have also successfully use stunnel (SSL) and SSH to accomplish
the same thing. I personally avoid SSL altogether.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Wednesday, May 02, 2007 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VPN between Asterisk server and phone client
Concur with Steve: OpenVPN is your friend. At one time, I used VPN on
Demand-type functionality in my dial plan to trunk a certain subset
of calls to a different * server via OpenVPN. This is what that
dialplan looked like:
[trunkfreecallsviaoffsite]
exten = _X.,1,NoOp
exten = _X.,n,Playback(creating_vpn)
exten = _X.,n,System(/usr/local/bin/startvpn clientname ${CALLERID(name)})
exten = _X.,n,Wait(10)
exten = _X.,n,Playback(success_vpn)
exten = _X.,n,Dial(IAX2/vpnmaster/**${EXTEN},60,TW)
exten = _X.,n,Hangup
exten = h,1,System(/usr/local/bin/stopvpn clientname ${CALLERID(name)})
exten = h,n,Playback(stopping_vpn)
The startvpn and stopvpn scripts (which I've since managed to lose)
would establish the VPN between this server and the vpnmaster
server. The scripts would also keep track of current users
(${CALLERID(name)} of the VPN-trunk. As a side effect of user
tracking, I'd know when the VPN was already established, so I didn't
need to re-connect. Similarly, I'd only tear it down when no users
were left.
As I mentioned, this does not address your direct need to create a VPN
between an endpoint (softphone) and your server. My example simply
illustrates the straight-forward OpenVPN approach. You can install the
OpenVPN GUI tools on your desktop/laptop and create the VPN manually
when you need it.
BTW, I stopped using this technique when we added a second local
server, so I didn't have to go across the WAN for offloading certain
calls anymore.
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RE: [asterisk-users] allowing call every 15mins
I don't think he only wants to receive calls once every 15 min. I think he
wants you not to be able to call back unless you wait 15 minutes. I guess he
doesn't have an ACD or a queue.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, May 02, 2007 4:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] allowing call every 15mins
On Wednesday 02 May 2007 3:04 pm, Goke Aruna wrote:
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is 15, 30, 45, 60(in multiples of 15 minutes).
So YOU're the guy who makes the calls to tech support so hideous!!
how can i achieve this and what application can i use to get this done.
GotoIfTime can help you here, but it'll be a little messy:
exten = 1,1,GotoIfTime(0:01-0:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:16-0:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:31-0:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(0:46-0:59|*|*|*?toobad)
exten = 1,1,GotoIfTime(1:01-1:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:16-1:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:31-1:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(1:46-1:59|*|*|*?toobad)
...
exten = 1,1,GotoIfTime(23:01-23:14|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:16-23:29|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:31-23:44|*|*|*?toobad)
exten = 1,n,GotoIfTime(23:46-23:59|*|*|*?toobad)
exten = 1,n,Dial(SIP/techsupport)
exten = 1,n,GotoIf($[${DIALSTATUS} = BUSY]?toobad)
exten = 1,n,Hangup
exten = 1,n(toobad),VoiceMail([EMAIL PROTECTED])
Very messy. Alternatively:
exten = 1,1,GotoIfTime(0:00-0:00|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:15-0:15|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:30-0:30|*|*|*?woohoo)
exten = 1,n,GotoIfTime(0:45-0:45|*|*|*?woohoo)
...
exten = 1,n,GotoIfTime(23:00-23:00|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:15-23:15|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:30-23:30|*|*|*?woohoo)
exten = 1,n,GotoIfTime(23:45-23:45|*|*|*?woohoo)
exten = 1,n,VoiceMail([EMAIL PROTECTED])
exten = 1,n,Hangup
exten = 1,n(woohoo),Dial(SIP/techsupport)
...
Pretty much equally messy.
Both of these examples assume you want to allow calls for a minute every
quarter hour 24 hours a day, quite possibly to match policies on most
vendors
which claim they offer 24-hour tech support but implement similar
dialplans. :-)
Honestly though this is a strange request... Why bother offering tech
support
if you are only allowing calls for 1 minute every 15 minutes? Why not be
honest about it and do this:
exten = 1,1,Playback(sorry-we-dont-offer-support)
exten = 1,n,Wait(30)
exten = 1,n,Hangup
??
-A.
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RE: [asterisk-users] VPN between Asterisk server and phone client
If you run it on the fly, doesn't that mean that the Asterisk user will have
permissions to configure VPN's? Nobody sees a problem with that? I thinking
that if you knock over the Asterisk service and get shell execution rights
as Asterisk, you could be able to start tunnels for things other than voice.
It's like giving a hacker a great way to hide their activities from your IDS
without having to bother to get root first to install an encrypted data
pipe.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, May 02, 2007 4:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VPN between Asterisk server and phone client
Kai-Uwe Jensen wrote:
Concur with Steve: OpenVPN is your friend. At one time, I used VPN on
Demand-type functionality in my dial plan to trunk a certain subset
of calls to a different * server via OpenVPN. This is what that
dialplan looked like:
[trunkfreecallsviaoffsite]
exten = _X.,1,NoOp
exten = _X.,n,Playback(creating_vpn)
exten = _X.,n,System(/usr/local/bin/startvpn clientname
${CALLERID(name)})
exten = _X.,n,Wait(10)
exten = _X.,n,Playback(success_vpn)
exten = _X.,n,Dial(IAX2/vpnmaster/**${EXTEN},60,TW)
exten = _X.,n,Hangup
exten = h,1,System(/usr/local/bin/stopvpn clientname ${CALLERID(name)})
exten = h,n,Playback(stopping_vpn)
The startvpn and stopvpn scripts (which I've since managed to lose)
would establish the VPN between this server and the vpnmaster
server. The scripts would also keep track of current users
(${CALLERID(name)} of the VPN-trunk. As a side effect of user
tracking, I'd know when the VPN was already established, so I didn't
need to re-connect. Similarly, I'd only tear it down when no users
were left.
As I mentioned, this does not address your direct need to create a VPN
between an endpoint (softphone) and your server. My example simply
illustrates the straight-forward OpenVPN approach. You can install the
OpenVPN GUI tools on your desktop/laptop and create the VPN manually
when you need it.
BTW, I stopped using this technique when we added a second local
server, so I didn't have to go across the WAN for offloading certain
calls anymore.
That is really a cool idea to add it on demand in the dialplan. Was the
wait(10) required to get the VPN up or could you set it to a lower
number? It seems OpenVPN connects pretty darn quickly. Did you ever
run into issues where wait(10) was not long enough?
Thanks,
Steve Totaro
www.asteriskhelpdesk.com
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RE: [asterisk-users] Calls in ulaw, not gsm as desired
Take a packet capture and look at the Polycom's SDP. It will be listed in the 'a=rtpmap' if iLBC is available. My bet is that you will only see: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Wednesday, May 02, 2007 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I see this. However, ulaw works fine. I believe alaw works as well. So I don't think their site is either correct, or they didn't mean it like that. Rob Jerry Jones wrote: A simple glance at their website will tell you this about the 501 G.711 μ/A and G.729A (Annex B) configuration On May 2, 2007, at 12:22 PM, Jaswinder Singh wrote: Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote: So I reloaded things and had just gsm set for 2 of my polycom 501 phones. However, the logs say No codec found, which leads me to believe that polycom 501 phones can't use gsm. Does anyone have something like this working? If not gsm, is there a better option with these phones over a low bandwidth situation? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] allowing call every 15mins
They could just put up a ticket system like Sellvoip and simply ignore all the tickets. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Wednesday, May 02, 2007 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowing call every 15mins Honestly though this is a strange request... Why bother offering tech support if you are only allowing calls for 1 minute every 15 minutes? Why not be honest about it and do this: exten = 1,1,Playback(sorry-we-dont-offer-support) exten = 1,n,Wait(30) exten = 1,n,Hangup ?? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ROFLMFAO!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 memory leak?
How much memory do you have on that server? Go ahead and install monit, configure it for checking memory usage, and have it page you when it goaes above 25%, etc. Only time will tell if it's a real memory leak that will need to be addressed. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, May 02, 2007 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 memory leak? You're right, 11megs isn't scary at all. It's the 106 megs from Monday that worried me. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Steve Finkelstein wrote: With all due respect, I believe you might be a bit paranoid. 10-11M is quite normal for the linux kernel to allocate for asterisk. It's not necessarily what the process is using, but that's just how memory management works within the kernel. What's 10-11M of RAM these days anyway? - sf Adam Moffett wrote: Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this afternoon I'm up to 11megs: VIRT RES SHR SWAP CODE DATA 30932 11m 560818m 1012 17m Is this a real issue or do I have something else going on? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VPN between Asterisk server and phone client
Any network service could potentially harbor a buffer overflow, etc that could result in remote command execution. Provided someone find a similar bug and it's exploitable, they would theoretically be able to spawn a shell with the same rights as Asterisk. Generally, it's better to run services as nobody. I would be hesitant to allow management of VPN's from within Asterisk. Check out this link: http://mixter.void.ru/exploit.html It's a basic tutorial on writing shell code for buffer overflows. The basic idea is you find some condition where you can cause the application to seg fault and if you are lucky, it will allow you to write your shell code to memory, gain control of the stack pointer, and make your shell code run. These types of exploits have to be tailored to specific OS's and architectures. Shellcode that works on a BSD system will not work on Solaris or Redhat, etc... Generally you can reuse the delivery code by swapping out the shell code for whatever you are attacking. I'm not stating these currently exist in Asterisk, but theoretically it is likely and we just don't know about it yet. Prudence suggest that we don't help the hackers any more than we have to in case they find it first. I think it would be really difficult to lockdown VPN if Asterisk manages it's operation. Asterisk would have to have execution rights to the VPN binaries or an intermediate script at the very least. Just my 2 cents. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, May 02, 2007 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VPN between Asterisk server and phone client On 5/2/07, Salvatore Giudice [EMAIL PROTECTED] wrote: If you run it on the fly, doesn't that mean that the Asterisk user will have permissions to configure VPN's? Nobody sees a problem with that? I thinking that if you knock over the Asterisk service and get shell execution rights as Asterisk, you could be able to start tunnels for things other than voice. It's like giving a hacker a great way to hide their activities from your IDS without having to bother to get root first to install an encrypted data pipe. That's true, the asterisk user needs to be able to invoke the start_vpn script or program. That does not mean that the asterisk user will have to have superuser rights to configure VPNs. You could make the start_vpn program setuid to a user that has those rights (and in that case, you probably don't want start_vpn to be a script). Also, openvpn typically starts predefined VPNs. To define a new one, someone would have to have access to the file system. When you say knock over the Asterisk servoce and get shell execution rights, how would that happen, exactly? I can think of DoS attacks and other stuff, but am wondering how knocking over Asterisk will give someone shell execution rights? As I said above, you would want to make the function to start a VPN connection as safe as possible. That would include NOT using scripts, and employing other verification methods. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
I have transitioned to other DID's. I think that company is out of business. Sellvoip is best avoided at all costs. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?
I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of people being hauled back to rehab. Anyway, maybe he just makes a habit of running off with people's money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, April 30, 2007 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip? On 2007-03-26 01:46:40 -0700, Salvatore Giudice [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. That seems like it was probably wise and I hope you followed through. I am now unable (for a week or so) to dial any outbound calls, or receive any at my did. Additionally when trying to call them at there local phone I get the disconnected message. They provided by FAR the best call quality for me when they where working, so I am going to miss them if they are gone forever. Also, I still have like 24$ (us) credit with them... I still hope they return, but wouldn't count on it. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] did we all get spammed by TechnoCo ?
That stuff is so dangerous. There are too many compliance requirements regarding spam. Doing this kind of stuff opens them up to a lawsuit in more than one state. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, May 01, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] did we all get spammed by TechnoCo ? Per Jessen wrote: I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. Ha -- I was just about to post something myself! Yes - I got this too, and immediately suspected a cull of addresses from the mailing list. I'm not impressed. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Change Codec
Put similar allow/disallow statements in the sip or iax entry you create for your outbound ip calls. Be aware that if you use different codecs for phones and your termination provider, all media will have to go through asterisk and you will incur the processing overhead of codec conversion. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arun Kumar Sent: Tuesday, May 01, 2007 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change Codec Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7940 no outgoing audio
You should get a packet capture of both cisco-cisco and grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be able to understand the other vendor's devices. BTW, what version of firmware are you running on the cisco phones? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Alman Sent: Tuesday, May 01, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7940 no outgoing audio Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent but the Cisco can receive voice from the remote phone fine. When we dial Cisco to Cisco it all works fine. I am at a loss to figure this out and any help pointing me in the right direction would be appreciated. We are running an old Asterisk server with version 1.0.10 (yeah we know) and the same mix of hardware and configs works fine. On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco firmware is 08-2-00. Any help appreciated. Regards Simon Alman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How many users can be supported simultaneously?
Will you be allowing reinvites? If the server processes media, it will obviously support less simultaneous calls. Also, you may want to rethink the wireless portion. Odds are you will have horrible QoS problems if you run multiple calls or mixed traffic over wireless. BTW, what do you use VPN for? Is this for remote access or for securing VoIP calls? If you are running SIP/G.711, you will need roughly 82.4 kb/s bandwidth. You won't have a bandwidth problem unless you're moving a massive amount of traffic through your VPN or web server. You will likely have a horrible QoS problem. My best guess is that you could push approximately 25 simultaneous calls with no codec conversion, but I wouldn't expect good quality audio. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonopoulos Angelos Sent: Tuesday, May 01, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How many users can be supported simultaneously? I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b I have checked in wikipedia but I did not find something Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Calls in ulaw, not gsm as desired
It's probably not your codec. Do you have your asterisk box on a Voice VLAN with priority queing configured? If you have mixed traffic on your uplink without VLAN's and priority queuing (or possibly 802.1p), then your QoS will suffer. Changing your codec to GSM will lower bandwidth consumption, but late packets are still late packets. If you can, try to get a measurement of latency to your peering provider before and after setting up QoS. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Calls in ulaw, not gsm as desired My problem is this We have a location outside of our network which is done over vpn. Everything works except for the voice quality to that location isn't very good. To try to resolve this, I wanted to try to make all calls go over gsm. Right now, when i say show sip channels, they all show ulaw signaling. My setup is pretty basic. I have realtime setup with mysql. In the sip.conf file, I did have disallow=all, allow=ulaw, allow=gsm. I removed those lines from there and in my sip_buddies table, I made sure that the extensions i'm using have disallow=all and allow=gsm. However, even once I reloaded the extensions, its still only using ulaw. Any thoughts? Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Calls in ulaw, not gsm as desired
Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Capture Asterisk traffic
I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] is dundi worth pursuing in this situation?
DUndi or enum only make sense if you plan to move extentions dynamically without having to touch you Asterisk configs or if you want to expose your addressing to the outside world. Personally, I would do it statically so you can avoid delays in processing addressing especially - in the case of enum- if you dns server becomes unavailable. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Tuesday, May 01, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] is dundi worth pursuing in this situation? At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again each with their own extensions and DID. Handling local and LD calls from all the offices isn't a big deal - just normal call routing for that. My main question is what to do with calls between the offices. Each branch is connected back to HQ with a persistant VPN tunnel - I've tested IAX2 traffic over these tunnels before, and things work great. Since this works fairly well, I envision using IAX trunks for all intra-office calls. So - in this situation, would it be easier to just manage the office dialplan(s) and call routing manually, or would it be worth it to set up dundi for extension discovery? Thanks! -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] T1 interface
You could get yourself a cisco universal gateway or a Audiocodes Mediant 1000 Single Span T1 SIP Gateway. With regard to the cards: In my experience, you want an echo cancellation card if you are connected to a carrier without echo cancellers. Typically, LEC circuits do not have echo cancellers and long distance carriers do. I personally do not buy Digium hardware anymore. I've had such an abysmal experience with Digium's hardware quality and overall support in th past that I now only use Sangoma equipment. I have never had a problem with Sangoma's equipment. Their service is exemplary and they have even offered me free professional services in the past to optimize my gateway setup. I wouldn't spit on Digium hardware if it was on fire. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: Tuesday, May 01, 2007 3:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] T1 interface Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207P vs. the TE205P? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Capture Asterisk traffic
Ethereal will let you export an rtp stream as a .au file. That's one of the very minor items we cover in our conference series and our VoIP 100 course. There is a lot more fun to be had when you get into RTP sequence number prediction and RTP stream I injection. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang Sent: Tuesday, May 01, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Capture Asterisk traffic I remember an app called 'vomit' that could allegedly reconstruct audio files from tcpdump pcap files. Salvatore Giudice wrote: I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port = 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Calls in ulaw, not gsm as desired
How did you set it to 5053? Can you post your sip.conf? You should remove the passwords and ip addresses, etc. Usually, it's just an allow and a disallow statement inserted into each inbound and outbound channel definition. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired That's what I did though. So my sip.conf file no longer has any allows in it. Instead, it should be relying on the realtime settings for that. However, even though I told it to only use 5053, it still is using ulaw. Rob Salvatore Giudice wrote: Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Stanaphone business ok?
Write them and ask. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd H Sent: Tuesday, May 01, 2007 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Stanaphone business ok? I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Applet?
If you want a commercial service, there are some decent companies out there like Estara. http://www.estara.com/ These services don't come cheap. Bluenote Networks also has a web CTC applet for their SIP PBX that they license to enterprises. http://www.bluenotenetworks.com They have a pretty nice VB/XML interface for a webserver to have call control over their PBX as well. If you want something free, you can try Jain-SIP: https://jain-sip-applet-phone.dev.java.net/ -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo L. Arturi Sent: Tuesday, May 01, 2007 6:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Applet? Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable applet for letting people call me directly with a single click. Anyone? Not sure if this question is off-topic, if so, please accept my apologizes. Thank you, Pablo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How many users can be supported simultaneously?
I would test that under load and see what kind of QoS you get. If you have mixed traffic streaming and non-streaming coming from the server, you will need 802.1p and an 802.1p compatible router. If possible, you may want to add a second wireless interface, bind asterisk to that interface, and have each wireless interface log into its own VLAN. Then you can use 802.1q and priority queuing to give the asterisk VLAN priority over the data VLAN when the remote access and webserver will operate. Everytime someone is on your VPN or hitting your webserver, your call quality will likely be impacted without a separate voice VLAN or 802.1p. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonopoulos Angelos Sent: Tuesday, May 01, 2007 6:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] How many users can be supported simultaneously? Thank you for the reply..VPN is used for remote access and for secure data transfer..The web-server does not have a lot of traffic..I use SIP and there is a grandstream gateway with 4 FXO..I think that 25 calls is a good number.. _ Από: [EMAIL PROTECTED] εκ μέρους Salvatore Giudice Αποστολή: Τρι 01/05/2007 19:14 Προς: 'Asterisk Users Mailing List - Non-Commercial Discussion' Θέμα: RE: [asterisk-users] How many users can be supported simultaneously? Will you be allowing reinvites? If the server processes media, it will obviously support less simultaneous calls. Also, you may want to rethink the wireless portion. Odds are you will have horrible QoS problems if you run multiple calls or mixed traffic over wireless. BTW, what do you use VPN for? Is this for remote access or for securing VoIP calls? If you are running SIP/G.711, you will need roughly 82.4 kb/s bandwidth. You won't have a bandwidth problem unless you're moving a massive amount of traffic through your VPN or web server. You will likely have a horrible QoS problem. My best guess is that you could push approximately 25 simultaneous calls with no codec conversion, but I wouldn't expect good quality audio. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonopoulos Angelos Sent: Tuesday, May 01, 2007 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How many users can be supported simultaneously? I have a pc with the following characteristics: Pentium IV 2.4Ghz HyperThreading 512 MB PC3600 Dual DDR RAM Seagate 80GB SATA HDD 4-port ethernet 10/100 PCI Card Netgear MA-311 802.11b Wireless Card On this machine runs a VPN server, an Apache server and an Asterisk Does anyone know or have experience about the number of users that could be supported for VoIP at the same time?It is a Wireless Lan over 802.11b I have checked in wikipedia but I did not find something Thanks attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digital Phones
Nortel digital Meridian phones are like $400/each. At least that was the price of the phones at a hotel I did a job for recently. When you go to SIP, you may save on the capital costs for the phones, but other costs will increase. These are related to: 1.) increased support requirements for supporting VoIP 2.) additional licensing may be required from your vendor to support SIP or IP media 3.) increased costs associated with mitigating potential damages since your voice services are now subject to the same outages as your network 4.) increased training costs for staff to become proficient in VoIP 5.) increased costs associated with monitoring QoS 6.) increased costs associated with reconfiguring your network for VoIP and QoS - many times new switches may have to be purchased in addition to SBC's etc 7.) additional costs associated with rewiring physical space to accommodate additional Ethernet ports required for phones Sometimes, if you already have digital - it may not be worth switching to SIP even if you save a ton on the handsets. Whenever we switch over a hotel to VoIP, we always run into these extra 'hidden' costs. If you want to do digital with asterisk, I think you'll need a T1/E1 multiplexer that supports digital phones. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, May 01, 2007 7:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Digital Phones What brand of digital phones, I think I read some time ago that someone was doing something with Nortel phones but I seem to remember the cost of the phone meant...better to toss the handsets and buy new sip handset. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tuesday, 1 May 2007 6:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digital Phones Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Applet?
You must be stoned. I have a client that has Estara service and only paid $2k USD for it. Im not going to show you anything. If you want to see what BNN has, you need to call BNN. They do have a web-based applet. Its based on Windows RTC. They have a lot more behind the counter than their SOA product. Pick up the phone and ask them. You need to do a bit more than just check out their website. I have no financial interest in selling Estara or BNN. I simply have worked with both these products before. If I had worked with your product, I would have given some information on it. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, May 01, 2007 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Applet? Salvatore, As Ive said before Estara charge $US50,000 minimum price for their solution. We discussed this last time, Im still waiting for you to show me a Click-to-Talk solution for anywhere near our price of £1,100 per server for the Corraleta SDK. Secondly I went to Bluenotenetworks, they dont have Click-to-Talk they only have click to dial you dont need to pay anyone for click to dial its a piece of cake you just use dynamically generate call files, if you go to Nerd Vittles he shows you how to set it up in about 10 minutes. Being able to dial out from an asterisk server to a web inputted phone number is very easy to do BUT its not what Pablo was after and nothing like Corraleta or Jiax. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation Call Button http://www.mexuar.com/ www.Mexuar.com Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Salvatore Giudice Sent: Tuesday, 1 May 2007 8:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Applet? If you want a commercial service, there are some decent companies out there like Estara. http://www.estara.com/ These services dont come cheap. Bluenote Networks also has a web CTC applet for their SIP PBX that they license to enterprises. http://www.bluenotenetworks.com They have a pretty nice VB/XML interface for a webserver to have call control over their PBX as well. If you want something free, you can try Jain-SIP: https://jain-sip-applet-phone.dev.java.net/ -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo L. Arturi Sent: Tuesday, May 01, 2007 6:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Applet? Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable applet for letting people call me directly with a single click. Anyone? Not sure if this question is off-topic, if so, please accept my apologizes. Thank you, Pablo image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 100 users - voip lan security and qos ?
This is a pretty common setup. Just make sure you have ACL's restricting traffic between your data and voice vlan's. Generally, we recommend more than two VLAN's for QoS and security. Usually customers setup the following: 1.) Voice VLAN's for Phones 2.) Data VLAN's for workstations 3.) Voice server VLAN's for IP telephony servers (anything that handles communications media) 4.) Data server VLAN's for intranet services 5.) converged communications VLAN's - Remote access VLAN's and workstation endpoints that have soft phones or IPTV clients fall into this category - 802.1p is recommended for these types of VLAN's 6.) wireless VLAN's - These are seldom built for QoS or streaming media, so they should be segmented and treated differently. All VLAN's should be properly segmented from each other. Ie. Data VLAN's should be restricted from accessing voice VLAN's. All network ingress/egress points should have appropriate SBC's and application layer gateways installed. The network should always be constructed to preserve voice services in the event of a network crisis. If you lose the data side of the network, 95% of large enterprises will always fall back on their telephone and conferencing systems for crisis management. Good luck. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Finkelstein Sent: Sunday, April 29, 2007 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 100 users - voip lan security and qos ? If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750) then you will be able to setup the phone and have the computer daisy chained to it. I have a similar setup on mine. Here's how I configure my switch ports in order to achieve the desired effect: switchport access vlan 5 switchport voice vlan 6 auto qos voip cisco-phone This is assuming your data VLAN is configured as VLAN 5, and your VoIP VLAN is on VLAN 6. This will allow the phone to create a trunk port and facilitate both end nodes through one switch port. HTH - sf A_ Navone wrote: i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own vlan ? 2 assuming the phone sends out a tos bit, how can we confirm that the customer's switch can read the tos bit and correctly prioritize it ? 3 to prioritize voip in the router (coming from the switch) we are looking at the wrtg54L and have found these 2 juicy websites http://openwrt.org and http://www.dd-wrt.com/dd-wrtv2/index.php has anyone downloaded and flashed the voip firmware ? does it give worthwhile advantages over the default firmware ? does the wrtg54L have any advantages over other routers ? any other advice to offer ? thank you so much in advance _ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglineapr il07 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,4634f9c388295209328925! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ADSL routers with integrated SIP QoS for other devices
Try the intertex gateways http://www.intertex.se/ Here their page outlining the their QoS settings: http://www.intertex.se/products/page.asp?iPageID=143 They have models with ADSL models and wireless access point components. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Saturday, April 28, 2007 11:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ADSL routers with integrated SIP QoS for other devices Greetings list, Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which I'll never visit (minimum 3 boxes - ADSL modem, m0n0, WiFi AP). So, does anyone have any recommendations for a wireless ADSL router with integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. Vigor 2700V), but I can't find reference as to whether the integrated QoS applies only to the FXS ports in the router itself, or to all SIP traffic (most of the users will have separate SIP hardphones). These are all to be used in the UK, so the device in question needs to support PPPoA. Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
I think you are assuming that the company owns the computer that the employee runs the soft phone on. It's possible that employees will want to run them from untrusted computers at home, etc. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, April 20, 2007 9:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote: Tzafrir, I don't know if you do many large deployments, but this would be a godsend! We did an install for a large law firm with all lawyers wanting softphones (eyebeam) on their laptops. Centrally pushing out the install executable was easy, setting up the parameters for each user was time consuming (i.e. expensive). With hard phones, we setup a TFTP server for each phone to pull config on bootup. We've even built a couple of tools to build config files (text ini files) dynamically from a database. This has shaved up to 8 hours off a large install. I think you're confusing installation with configuration. Without ascii config files (or a tool from the mfg to create binary config files from a script), each soft device must manually configured. I am not. The soft phone is not the only software on that computer that needs cetral configuration. How do you configure the networking on those computers? The mail clients? How do you deploy updates? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
If it's a law firm, they are probably using Windows. I believe the initial post mentioned they were using Counterpath as well. BTW, if you are using X-ten, you can script a launcher application which can perform your provisioning download/authentication and provision the client by setting the appropriate registry entries. The part that sucks with Counterpath is that it's difficult to generate the encrypted string they use to store the password in the registry key. The work around is to generate sample passwords and capture those from the registry. Use the plain text password in your sip service and set the client to the encrypted string with the launcher script. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, April 20, 2007 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Fri, Apr 20, 2007 at 09:21:54PM -0400, Michelle Dupuis wrote: Tzafrir, I don't know if you do many large deployments, but this would be a godsend! We did an install for a large law firm with all lawyers wanting softphones (eyebeam) on their laptops. Centrally pushing out the install executable was easy, setting up the parameters for each user was time consuming (i.e. expensive). With hard phones, we setup a TFTP server for each phone to pull config on bootup. We've even built a couple of tools to build config files (text ini files) dynamically from a database. This has shaved up to 8 hours off a large install. I think you're confusing installation with configuration. Without ascii config files (or a tool from the mfg to create binary config files from a script), each soft device must manually configured. Can you name a decent Linux soft phone worth its salt for which you cannot generate such a provisioning system in 1 hour? It would probably be custom and site-specific. The generic provisioning systems used for soft phones require way too much trust on the provisioning server and are lacking on the security side. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Java require 26 28 times more processing resources than C/C++. Perl is about 1.2 times vs. C/C++. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, April 21, 2007 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Softphone that supports central provisioning? Lol, yep you missed something but do you really want to be taught something you already think you know? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation Call Button http://www.mexuar.com/ www.Mexuar.com Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 21 April 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? What's your objection to a softphone in java ? Java is slow and the interface is always ugly and doesn't fit into the window manager etc. you are used to. :-P I never understood why I would use Java to write software when I could use C(++) or when a script language would do. The simple fact that people have 2 or 3 GHz doesn't mean that I have to burn them for nothing. The only point may be portability. Do I miss something? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ image001.gif Description: GIF image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Software vendors like java because it gives them easy access to a large pool of cheap labor fluent in Java and it also means not having to maintain more than one code base. C/C++ programmer are becoming more scarce every year since most colleges and universities start with Java now instead of C/C++. Generally, it's fine for GUI's or client side processing, but it is not fine for any application which requires performance. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Saturday, April 21, 2007 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? For things running inside the browser, i think java is a reasonable choice. Yes you could do it with active-x too, but it won't work on all OS'es. I hate java, probably for the same reasons you do, but in same cases its the best option. Zoa Dean Collins wrote: Lol, yep you missed something but do you really want to be taught something you already think you know? Regards, Dean Collins [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation ** www.Mexuar.com ** http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 21 April 2007 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? What's your objection to a softphone in java ? Java is slow and the interface is always ugly and doesn't fit into the window manager etc. you are used to. :-P I never understood why I would use Java to write software when I could use C(++) or when a script language would do. The simple fact that people have 2 or 3 GHz doesn't mean that I have to burn them for nothing. The only point may be portability. Do I miss something? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Most large enterprises (25k+ employees) would rather have a product backed by a real vendor, are not willing to switch office workers to linux, and do not see IAX as a viable option. Many customers avoid things like IAX form fear of being tied to a single vendor. I don't think it's generally possible to dismiss products that don't meet all of your business requirements. It would probably be a better idea to find a few candidates that meet most of your business requirements or at least the higher priority requirements, and then work with the vendor's professional services staff or your own development staff to tailor the product to your needs. None of the products you mentioned would ever be acceptable to the majority of enterprise clients I work with. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, April 21, 2007 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Sat, Apr 21, 2007 at 02:00:05AM -0400, Salvatore Giudice wrote: If it's a law firm, they are probably using Windows. I believe the initial post mentioned they were using Counterpath as well. BTW, if you are using X-ten, you can script a launcher application which can perform your provisioning download/authentication and provision the client by setting the appropriate registry entries. The part that sucks with Counterpath is that it's difficult to generate the encrypted string they use to store the password in the registry key. The work around is to generate sample passwords and capture those from the registry. Use the plain text password in your sip service and set the client to the encrypted string with the launcher script. You asked if we knew a specific softphone that can be provisioned. Well, if it isn't configurable enough, it is not good enough. Then you should not use it. Use twinkle. Use kiax. Use iaxcomm. Just don't don't complain that this specific software is not configurable enough. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
If they are already using eyebeam, I would suggest contacting Counterpath and ask them to give them a branded client and possibly a launcher script that does an HTTPS POST. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, April 21, 2007 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Softphone that supports central provisioning? Last time I checked the official stats 88.5% of all browsers have Java 1.4 or above installed. Therefore apart from the 125kb Corraleta applet 88.5% of browsers can be making calls in 10 seconds or less from downloading to dialing for the first time. Sounds pretty reasonable to me. But like I said I don't think you are asking the question because you want to learn something. People have suggested alternatives but none seem 'suitable' for you. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation Call Button http://www.mexuar.com/ www.Mexuar.com Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, 21 April 2007 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone that supports central provisioning? That it still requires installation of a JVM on the client's system (most browsers nowadays don't have it installed). And then again, it required downloading the software to be used from the server, which is not such a grand idea. If Java were installed by default in the browser, it would be nice to use such a phone. But as things stand now, a simple softphone is just as good. Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir image001.gif Description: GIF image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
Can you receive calls on Corraleta or is this just another Click-to-call web applet? One of the hotels I worked with successfully used a similar product from Estara. http://www.estara.com/ If you are just looking for click-to-call, then check out Estara as well. I thought the original poster was looking for a full provisioned soft phone client not just click to call. I believe he mentioned wanting to manage softphones like deskphones, which would imply receiving calls as well. Obviously this could be faked with ec500 or similar call forwarding to a home device. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Saturday, April 21, 2007 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Sat, 21 Apr 2007, Philipp Kempgen wrote: Tim Panton wrote: What's your objection to a softphone in java ? Java is slow and the interface is always ugly and doesn't fit into the window manager etc. you are used to. :-P I never understood why I would use Java to write software when I could use C(++) or when a script language would do. The simple fact that people have 2 or 3 GHz doesn't mean that I have to burn them for nothing. The only point may be portability. FWIW: I've been trialling the Mexuar Java phone over the past few days, and I feel that I have to say that what you've just written really doesn't apply. So what if you have to burn the cpu and need a 2GHz processor? Here the the output from top on my desktop when it's running: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 5211 gordon15 0 272m 31m 14m S 5.3 4.2 0:03.59 java_vm CPU details from /proc/cpuinfo: model name : AMD Athlon(TM) XP 2200+ cpu MHz : 1800.231 cache size : 256 KB So it's hardly a new processor, and 5% usage is nothing. So Java is far from slow these days. As for the interface - it's as ugly as you care to write a web-page round it. Mexuars own one looks pretty good to me, and it's 100% customisable. The workhorse is very cleverly hidden behind a standard web-page and javascript interface, so it's you who writes the interface and javascript shim to interface to the java applet, not the vendor (unless you pay them, I guess ;-) You made the point of portability - a big plus for me. My desktop is Linux, but I work with people who have Mac and Win desktops. Having something that looks the same and acts the same over all platforms is a boon (principle of least surprise) Finally, Java is doing what it was always meant to do, and people are starting to understand this too. (and I'm not personally a fan of Java either and I was skeptical when I saw this, but it does exactly what it says on the tin when used in this manner) Do I miss something? I think you're missing a great opportunity. And one other thing - you don't have to write anything other than some web page in html and javascript - Mexua have written the hard bits for you, and licensing costs are on-par with getting a custom idefisk or x-lite. Remote provisioning of this in an office (or in home offices if you out-house your agents) can be trivially done by having the web server serve up different pages for each client. Something easy to do on IP address, or based on the agents login to the web system, you can customse the front-end, so no call buttons are visible, or no dial window. All the agent does is wait for the phone to ring and hit the answer button. (sucks to be an agent though ;-) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
I don't think that pricing is accurate for Estara. I have seen a deal for a Boston hotel that was significantly cheaper. BTW, how many enterprise-class customers have you won Asterisk sales by using Correlata? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, April 21, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Softphone that supports central provisioning? Yes you can receive calls on it well, please see the Tesco phone example. Yes it is similar to estara the worlds largest click to call technology who has customers such as Hilton hotels BUT estara is active-x so restricted on which OS and estara starts!!! At $US50,000 license fee. As you can see from here http://www.voip-info.org/wiki/view/Mexuar the licensing restrictions are very favorable to Asterisk system integrators. My job for Asterisk integrators in the Americas, Asia and Australia (and Charles my counterpart in the UK - who looks after Europe, Middle East and Africa) is to assist you guys in closing Asterisk sales by implementing the Corraleta technology. There are a number of uses of this technology (eg www.Mexuar.com/Demo/Demo4 ) that will help you win Asterisk sales that cant be delivered via Cisco/Nortel etc and I'm here 24x7 as a resource to make you guys successful. Anyway getting a little bit commercial this discussion and should probably be moved to the Biz list, but if anyone has any questions or would like any more information call me on the numbers below here in New York to help. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation Call Button http://www.mexuar.com/ www.Mexuar.com Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Salvatore Giudice Sent: Saturday, 21 April 2007 1:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Softphone that supports central provisioning? Can you receive calls on Corraleta or is this just another Click-to-call web applet? One of the hotels I worked with successfully used a similar product from Estara. http://www.estara.com/ If you are just looking for click-to-call, then check out Estara as well. I thought the original poster was looking for a full provisioned soft phone client not just click to call. I believe he mentioned wanting to manage softphones like deskphones, which would imply receiving calls as well. Obviously this could be faked with ec500 or similar call forwarding to a home device. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 image001.gif Description: GIF image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
I tried to get the SJPhone folks to implement this two years ago. It's one of the major features missing from the market. You may want to contact Bluenote networks. http://www.bluenotenetworks.com/ They have an IP PBX and a soft phone client. They only sell their products to the enterprise market. They can do this for you. Their clients and servers are based on the Microsoft RTC and Radvision stacks. If you have any problems, tell them I referred you. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Friday, April 20, 2007 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Softphone that supports central provisioning? Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has soft keys (yes, I know they're all soft - but you know what I mean; programmable buttons whose function comes from the provisioning system), even better. I know idefisk Biz says they'll do this, but it's not in the release candidate and will make it's debut in the final version, which is a little too much early adoption for my liking. Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in binary files, preventing me from I'm open to SIP/IAX, so long as I don't have to jump through hoops to get it talking to *. Thanks for any experience you can share. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
A complete provisioning system for soft phones could impart some of the same authentication models used for popular IM clients. Imagine a large enterprise who wants to give out several thousand soft phones to employees in a turnkey fashion requiring the employee's network credentials to authenticate at the start of each session. Generally, it is not acceptable to use employee credentials to perform SIP digest authentication. Employee credentials are meant for employees, not devices or software that sets up a session on behalf of an employee. The solution to this kind of setup is to use a soft phone that can be downloaded on demand and presents the employee with a simple username/password/domain login box. In one such system that I worked on, the client would take the credentials from the employee and authenticate via HTTPS to a simple CGI script that authenticates the credentials against an Active Directory setup. Once the employee is authenticated, the CGI script sets a temporary password in a database that is accessible by a radius server and sends back all the provisioning information including the employee's office number and the temporary session password via XML in the HTTPS POST response. The client then logs into the SIP service using the session credentials. The employee is required to re-authenticate at the start of each soft phone session or after a timed interval when the temporary session password is expired from radius. The advantages to this kind of setup are: 1.) you don't have employee credentials stored in soft phones 2.) you avoid locking out employee credentials when policy-based password changes are required because of rapid authentication failures from a SIP device with stored credentials 3.) no SIP service credentials are stored in the soft phones 4.) in the event that the temporary session password is stolen from a soft phone installation, it is only good for a short period of time usually limited to 12 hours 5.) HTTPS is a significantly better provisioning method than TFTP (cough Cisco...) because it is encrypted and you have the opportunity to validate a cert from the provisioning server to ensure that the soft phone client is talking directly to the provisioning server. Man in the middle attacks suck. 6.) it's a lot easier to change provisioning information for all clients without requiring employees to download a new soft phone with hardcoded settings or trying to get employees to implement changes on their phones manually. For the same reason, it reduces initial setup complexity and also eliminates the bulk of setup related support calls We have put together implementations of this kind of system before for clients. Usually, this kind of scenario is not something we discuss outside our training classes or at conventions. Generally, this kind of system is commonly requested by enterprise and government customers when they seek to extend their phone system to employees for road warrior, pandemic, disaster recovery, or occasional work at home scenarios. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, April 20, 2007 9:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? Why would you want to do that? There are well-known and established tools to provision (centrally configure) software running on computers in a entwork. Why should the soft phones be configured any differently? What OS do you use on the desktops? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Softphone that supports central provisioning?
I hope to god you didn't put that TFTP server on the open internet. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Friday, April 20, 2007 9:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Softphone that supports central provisioning? Tzafrir, I don't know if you do many large deployments, but this would be a godsend! We did an install for a large law firm with all lawyers wanting softphones (eyebeam) on their laptops. Centrally pushing out the install executable was easy, setting up the parameters for each user was time consuming (i.e. expensive). With hard phones, we setup a TFTP server for each phone to pull config on bootup. We've even built a couple of tools to build config files (text ini files) dynamically from a database. This has shaved up to 8 hours off a large install. I think you're confusing installation with configuration. Without ascii config files (or a tool from the mfg to create binary config files from a script), each soft device must manually configured. Michelle Dupuis Technical Support Specialist Generation Software - Linux and Asterisk solutions and support. Visit us at www.generationd.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, April 20, 2007 9:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone that supports central provisioning? On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? Why would you want to do that? There are well-known and established tools to provision (centrally configure) software running on computers in a entwork. Why should the soft phones be configured any differently? What OS do you use on the desktops? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I'm sure they are exploring all options. Eventually, it's just a matter of time until the investors start with the class action lawsuits. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, April 17, 2007 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Just saw this article this morning: http://www.engadget.com/2007/04/16/vonage-no-workaround-were-pretty-much-scr ewed/ What happened to their workaround, whereby they route all of their traffic to someone else, who takes cares of LCR and ENUM? I don't understand how that wouldn't indemnify Vonage. On 4/13/07, Salvatore Giudice [EMAIL PROTECTED] wrote: My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702)979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendations for a voip provider who supports LNP?
(sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Recommendations for a voip provider who supports LNP?
I need a straight origination/termination provider on a per minute charge plan. I would like to avoid a monthly subscription-based provider. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, April 17, 2007 6:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Recommendations for a voip provider who supports LNP? On 4/17/07, Salvatore Giudice wrote: (sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG have you considered teliax.com ? check your numbers for LNP at the bottom left. I have been playing with voip for only about a month, but no complaints with teliax svc so far. -baji. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HTTP Connection Timeout Trouble with Cisco 7960 Phone
You should run a packet capture. Verify that a request is sent to the server and then verify if you see a response come back. I assume you have a cgi that generates the xml needed to display on the phone. So either, your cgi is not responding or the response is not formatted properly for the phone. Also check to make sure that your http proxy settings are correct in the phone. It's possible that you have a proxy set that is incorrect or that you need to set one in order to get to your webserver. These types of scenarios can cause the problem you described. Good luck. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hennessy Sent: Saturday, April 14, 2007 12:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] HTTP Connection Timeout Trouble with Cisco 7960 Phone Hello, I'm using two Cisco 7960 phones currently loaded and showing Firmware POS3-07-4-0 (Version 7.4?) and I'm having a strange problem. Whenever the phone is supposed to try to load anything over HTTP from my Apache 2.2.x web server, the connection just sits and times out. Nothing shows up in the Apache logs unless I hit cancel. What could the trouble be? -- Mark P. Hennessy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Job listing on cisco.com for Asterisk...?
Roflol. The chance of that happening are slim to none. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, April 13, 2007 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Job listing on cisco.com for Asterisk...? NewsFlash Cisco Acquires Digium for $1.4 Gazillion dollars Mark Spencer seen flying off in a lear jet en-route for Barbados. * Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Friday, 13 April 2007 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Job listing on cisco.com for Asterisk...? I thought this was interesting, if you are in China and need a job, you might also... http://www.cisco.apply2jobs.com/index.cfm?fuseaction=mExternal.showJobR ID=7 71671CurrentPage=1 * Working knowledge : Asterisk PBX; SIP Proxy Servers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Which SIP phones to buy?
Product selection is not cut and dry. What are your business requirements? So you need encryption? If so, what kind? Do they need support for outbound proxies? Are you going to use the same model for remote deployments? Do you need WAP capabilities? Do you need programmable speed dials? Do you need modular admin sidecars? Do you need IPSEC capabilities built into the handset? Do you need advanced/specific codec support? Do you need guaranteed interoperability with specific vendor supplied components? Are you looking for a phone for 10 people, 100 people, or 1 people? If you are scaling, what does your provisioning system look like? Do you need phone features like video or quality speaker phone? What is your budget for phones? Do you need an RTCP capable handset? Do you need a handset that support 802.11p for QoS? The more specific you can get about your business requirements, the better targeted your product selection will be. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, April 11, 2007 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which SIP phones to buy? I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
My wife's name is Nanae... =) The VoIP patent stuff is something that needs to be talked about more. VoIP is really going to suffer in the years to come because of patents. Might make a good topic for a whitepaper at a conference of speaking engagement. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Friday, April 13, 2007 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit I love this thread, especially when it came to the chicken boner part of the discussion - brings back NANAE with a smile - and I'm glad no one found it off-topic, I think it's well worth talking about (the suit, not the chicken boners) as this may have an effect on some of what we do. I hope to talk a little about it on the Asterisk Users Conference today at 12:30 EDT if anyone wants to. Otherwise, it's about features.conf and whatever else comes up. For info, see http://x2z.eu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
You hit the nail on the head. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, April 11, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemming=on It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) The US patent system is totally broken. It started with lobbying efforts to relax the applicability rules for patents for short-term gain. In the long term, it's going to do big damage to American competitiveness. And that's the sad thing about this. It discourages actual innovation (despite Wall Street protests to the contrary). If everytime you want to build on somebody else's work you have to build a skein of licencing agreements, you start to ask yourself, why should I bother? More and more companies are answering that one with We shouldn't -- there's enough action to be had in other parts of the world, where the conditions are much less onerous. Another example of that kind of short-sighted thinking is what happened to the US crypto business when all the export controls were brought in. (A lot of damage was done in exchange for no demonstrable security benefit.) Obviously, a market that big and moneyed isn't going to be ignored: how can it be? But what used to be a no-brainer isn't so obvious anymore -- staying out of the US market is a serious option where it wasn't before, and that just leads to further Balkanization. It's fitting that an open source product like Asterisk is helping keep the US in the game. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Coccimiglio Sent: Tuesday, April 10, 2007 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit These are the patent numbers in the lawsuit (Thanks Pat and Sal) 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 Mark C. Yuan LIU wrote: From: Kenneth Padgett [EMAIL PROTECTED] Date: Mon, 9 Apr 2007 23:49:31 -0400 [good stuff sniffed] I'm not doubting that patents exist, I'm just betting that you'd have to have some seriously drunken vision to interpret them as the exact business processes Vonage uses. I think if Verizon thought for a second they had solid ground to stand on, they would disclose which patents they're referencing so the public could decide. I bet you can access court records under some public information access laws. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I co-invented that one. It's a good one. A lot of my input went into it, but the final product was much more general than I was originally led to believe. Somehow this patent was slowly changed from disclosing sip contact center technologies to a patent on using SIP. The original intention was to do this for disclosure purposes in order defend against clowns like Katz. However, the company that owns this patent has since transferred rights to one of their subsidiary IP PBX firms and eventually they may decide to use this patent for other purposes besides defensive disclosure. I imagine that they could always whip this patent out on competing SIP PBX companies... It certainly would be annoying to deal with. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, April 10, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemming=on It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, April 10, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: Take a look at this patent: http://www.freepatentsonline.com/20060098624.html Title: Using session initiation protocol Document Type and Number:United States Patent 20060098624 Inventors: Morgan, David P. (Lexington, MA, US) Sullivan, Daniel B. (Charlestown, MA, US) Erickson, Jon A. (Scituate, MA, US) Giudice, Salvatore R. (Charlestown, MA, US) This is the kind of stuff that goes on in corporate America when it comes to new technology and patent law. =) Holy cow. -Stephen- You ain't kidding!!! Next thing you know someone will try to patent this: User picks up communications unit human interface device, a.k.a. 'handset', in response to audible ringing indication (visual 'ring' indication is optional). Just when I thought I couldn't have a lower expectation for a government agency - here comes the USPTO. Monumental foolishness. -MC P.S. - in broader terms, are there any of these patents that threaten FOSS telephony projects? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I think it's a small, feather covered appendage. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Monday, April 09, 2007 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Christopher Chan wrote: Welcome to China. Most Hong Kong'ers loathe mainland chinese and if they could, they would never get a job in China. I get plenty of hack attempts too from China however I doubt that is due to the same sentiment in China. If you want to find someone to blame, please look no further than the US where your chicken boners are in league with crackers and virus writers to create botnets to send their spam. This is of course besides the ignorance of those who own computers in China (man, computers there are infested with virii, worms and trojans) that run that most secure of operating systems Microsoft Windows and those who actually get paid by chicken boners to host their crap. Oh, there are plenty of hack attempts from Korea too. Are you going to add Korea to the list of 'IP' violators too? Just curious, Christopher, what is a chicken boner? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon Vonage 101
They could be suing for patents completely unrelated to VoIP as a technology. There are cases on the book where people like Katz have been running around suing contact center operators because he has a patent on authenticating yourself to a phone service using a pin number and using that information to access account records. Contact center operators get hit by that crap all the time and Katz is a master of letting them operate for years before he comes banging on their door looking for a check. With VoIP, the news is always talking about subscriber numbers and industry growth projection. This is like putting blood in the water. It was only a matter of time until the guys like Verizon, Katz, etc start pulling their polished patent infringement weapons out on the naive VoIP operators. This is just how business is done in America. You want to see a whacked out patent? Take a look at Katz's patent on Methods and apparatus for intelligent selection of goods and services in telephonic and Electronic Commerce. This guy has patents on paying by phone or web for products using a credit card. There are 267 different methods this clown has patented and he actively sues companies for using these methods in common business channels. http://www.google.com/patents?vid=USPAT6055513id=VGQEEBAJ At my former employer, when VoIP was starting to get hot - they had me apply for a patent on IP contact center technologies which took a lot of what Katz had produced and expanded it to VoIP. We did this for purely defensive disclosure purposes, but there are clowns out there who do this to generate revenue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo Sent: Saturday, April 07, 2007 11:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Verizon Vonage 101 I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant messenger US 7,054,308 B1 Method and apparatus for estimating the call grade of service and offered traffic for voice over internet protocol calls at a PSTN-IP network gateway According to Google: They've listed 118 patents assigned to Verizon Results 1 - 10 of about 118 from uspto.gov for +assigned to +verizon One dealing with PSTN Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon +pstn Two matches dealing with VoIP but only one is a patent. And that is related to the above search Results 1 - 2 of 2 from uspto.gov for +assigned to +verizon voip Three matches dealing with Voice and IP but only one is a patent. And that too is related to the above search Results 1 - 3 of 3 from uspto.gov for +assigned to +verizon voice IP Nine matches dealing with telephone and IP but only one is a patent. And that too is related to the above search Results 1 - 9 of 9 from uspto.gov for +assigned to +verizon telephone IP. One patent related to voicemail Results 1 - 1 of 1 from uspto.gov for +assigned to +verizon voicemail My thoughts, the voicemail one is broad, and could be circumvented easily. If I were a juror, I would laugh but an infringment is an infringement is an infringement. I would make Vonage stop using the technology. The VoIP patent however is a bit more detailed, and although it can be construed as broad, that too would make me side with Verizon, but not to the degree of shutting down Vonage. On the flip side of things, Vonage is no stranger to infringing on patents. Of course, turnabout is fair play, and Klausner Technologies Inc. filed suit against Vonage for infringing its patent number 5,572,576, which concerns the retrieval of VoIP voicemail on a cell phone or handheld device. http://www.cedmagazine.com/article/CA6351074.html -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
RE: [asterisk-users] Verizon Vonage 101
Here's links and descriptions for the 8 you listed. All Bell Atlantic, GTE, or Verizon. This should make your research a bit easier. 6,137,869 Network session management http://www.google.com/patents?vid=USPAT6137869id=yl4GEBAJdq=6137869 Patent number: 6137869 Filing date: Sep 16, 1997 Issue date: Oct 24, 2000 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Network Services, Inc. Primary Examiner: Rexford N Barnie 6,430,275 Enhanced signaling for terminating resource http://www.google.com/patents?vid=USPAT6430275id=NmwLEBAJdq=6,430,275 Patent number: 6430275 Filing date: Jul 28, 1999 Issue date: Aug 6, 2002 Inventors: Eric A. Voit, Edward E. Balkovich, William D. Goodman, Jayant G. Gadre, Patrick E. White, David E. Young Assignee: Bell Atlantic Services Network, Inc. Primary Examiner: Curtis Kuntz Secondary Examiner: Rexford M Barnie 6,104,711 (The famous: We think we invented ENUM patent) Enhanced internet domain name server http://www.google.com/patents?vid=USPAT6104711id=J18EEBAJdq=6,104,711 Patent number: 6104711 Filing date: Mar 6, 1997 Issue date: Aug 15, 2000 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,282,574 Method, server and telecommunications system for name translation on a conditional basis and/orto a telephone number http://www.google.com/patents?vid=USPAT6282574id=46sIEBAJdq=6,282,574 Patent number: 6282574 Filing date: Feb 24, 2000 Issue date: Aug 28, 2001 Inventor: Eric A. Voit Assignee: Bell Atlantic Network Services, Inc. 6,359,880 Public wireless/cordless internet gateway http://www.google.com/patents?vid=USPAT6359880id=tP4KEBAJdq=6,359,880 Patent number: 6359880 Filing date: Jul 30, 1999 Issue date: Mar 19, 2002 Inventors: James E. Curry, Robert D. Farris Primary Examiner: Wellington Chin Secondary Examiner: Steven Nguyen 6,128,304 (We think we own presence too...) Network presence for a communications system operating over a computer network http://www.google.com/patents?vid=USPAT6128304id=BnkGEBAJdq=6,128,304 Patent number: 6128304 Filing date: Oct 23, 1998 Issue date: Oct 3, 2000 Inventors: Steven E. Gardell, Barbara Mayne Kelly, Rajiv Bhatnagar, Thomas James Antell, Israel B. Zibman Assignee: GTE Laboratories Incorporated Primary Examiner: Frank Duong 6,298,062 (aka. Accepting H.323 phone calls/faxes from a computer network and terminating them on the PSTN) System providing integrated services over a computer network http://www.google.com/patents?vid=USPAT6298062id=jp4IEBAJdq=6,298,062 Patent number: 6298062 Filing date: Oct 23, 1998 Issue date: Oct 2, 2001 Inventors: Steven E. Gardell, Israel B. Zibman Assignee: Verizon Laboratories Inc. Primary Examiner: Shick Hom -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Buller Sent: Saturday, April 07, 2007 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon Vonage 101 J. Oquendo wrote: So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... I am looking at Verizon Services Corp. v. Vonage Holdings Corp., Slip Copy, 2007 WL 528749, E.D.Va.,2007, which is the result of the Markman hearing. That is the court interpreting the claim language, and here are the patents discussed: 6,137,869 6,430,275 6,104,711 6,282,574 6,359,880 6,128,304 6,298,062 I do not know which of these Vonage was found to have infringed. Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vonage fraud controls
Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in Asterisk and had your account terminated by Vonage? I'm curious as to whether they will stop your service if you push too many calls through their ATA in a specific period of time. Thanks in advance for the info, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Configuration assistance needed.
Where are you located? -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim King Sent: Thursday, April 05, 2007 11:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Configuration assistance needed. I have a fairly large system to configure. I was hoping to find someone locally to employ for this project but remote configuration is considerable. Pleas let me know if you are interested and have the time. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Require only GSM Codec
Not every client supports gsm. Usually it's a good idea to put ulaw as well or you could get errors when neither side supports the same codec. disallow=all allow=gsm allow=ulaw -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Zacarias Afonso Sent: Tuesday, April 03, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Require only GSM Codec Hi Sanjay, I'm not sure about that, but I think you can configure it in, for example, /etc/asterisk/sip.conf. There is an option that you configure for each channel like: only=gsm It instructs the sip protocol, that only gsm codec must be used. I hope it has helped you. Regards, Ronaldo. On 4/3/07, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Security on long distance calls
Using caller id to authenticate anyone is asking for a toll fraud problem. 4-digit pins really are not a good idea either. Try putting your operators and your users in different contexts. If you have specific numbers you don't want the users to be able to dial, then create patterns for those numbers. Play a sound byte that says the call is not allowed and then hang up. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, April 01, 2007 5:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Security on long distance calls Or you can do exten = _011.,1,Authenticate(1234) exten = _011.,2,Dial(SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED] ) Also this is a bit more complicated but you can do it by sip extension. If CID of phone = phone that is allowed then let it go out. This will be hard considering you will have to make a gotoif for every extension that you want to allow to call intl. - Original Message - From: Rizwan Hisham mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Friday, March 30, 2007 11:18 AM Subject: Re: [asterisk-users] Security on long distance calls For operator: [longdistance] include= local include= International for extensions: [localcalls] include= local now assign longdistance context to operator and localcalls context to every other user for whom you want to restrict intl calls [local] should include all local extension codes [International] should include all international extension codes you get my point? On 3/30/07, Stefano Corsi [EMAIL PROTECTED] wrote: Hello, which kind of method could you use to inhibit long distance calls to _some_ extensions? Is there a way to do it with freepbx or you have to do it manually in the config files? I wouldn't like to set a route password, because that is not confortable for the pbx operator. I just would like the operator being able to call whatever number, while the extensions should only be able to make local calls. Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)
Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will be a lot cleaner than taking a standard server distribution of linux and stripping out all the unwanted kernel modules. Monte Vista is an embedded linux that should be able to boot your server off a 128mb usb stick with Asterisk installed. You should probably strip asterisk down to the bare essentials for your project as well. You should be aware that flash memory is generally not the best medium to store data when you have a high number of read/writes. Flash memory will fail much more quickly under these conditions. You might want to consider using a usb microdrive instead of a flash stick. Pick a microdrive that generates as little heat as possible. BTW, what exactly is the motivation for running linux off of a usb stick? If you would like cdr's, you could likely do so with ngrep and a perl script. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Sunday, April 01, 2007 9:08 AM To: Asterisk-Users Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone) Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 31 Mar 2007 16:02:06 -0500 From: Mike Lynchfield [EMAIL PROTECTED] Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?
That's quite interesting. You can get the microdrives cheaper than $50. We recently purchased 2 GB microdrives for $17. Try contacting this company: IPMedia Asia Co. Ltd PO Box 2074 Northbrook Illinois United States 60065 Tel: (886 2) 85227000 Ext : 107 (1 847) 6565759 Fax: (886 2) 66021000 / (1 847) 5560164 IPMedia Asia Co. Ltd 10F-3, No. 107 Jhongshan Road, Sec. 1 Sinjhuang City Taipei Taiwan 24250 Tel: (886 2) 85227000 Ext : 107 Fax: (886 2) 66021000 One of our partner firms in Japan purchases USB sticks from them for promotional distribution at security conferences. They also have a line of Microdrives, I think you will find quite affordable. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] Sent: Sunday, April 01, 2007 1:47 PM To: Salvatore Giudice Cc: Asterisk-Users Subject: RE: On Topic: Cheapest Asterisk USB Key? I need a USB microprocessor *device* on which the Linux and Asterisk will run (even if very slowly), not just a storage drive from which to run it on the PC. MonteVista is a good distro, though there are other minimal embedded distros, of which I've already got one selected. The CDR usage of a single user's PC is just fine in performance and total lifetime read/writes (usually upwards of 100K) for the CDR data that needs to persist, as opposed to the device's RAM for executing the Asterisk. I'm looking for a device under $100 or $50 in OEM quantity, which is where just microdrives start. I want to run Asterisk itself, even if stripped down, for easy sync and single platform maintenance across all the Asterisk instances I've got, as well as guaranteed compatibility between data/network formats/protocols. On Sun, 2007-04-01 at 13:08 -0400, Salvatore Giudice wrote: Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will be a lot cleaner than taking a standard server distribution of linux and stripping out all the unwanted kernel modules. Monte Vista is an embedded linux that should be able to boot your server off a 128mb usb stick with Asterisk installed. You should probably strip asterisk down to the bare essentials for your project as well. You should be aware that flash memory is generally not the best medium to store data when you have a high number of read/writes. Flash memory will fail much more quickly under these conditions. You might want to conside using a usb microdrive instead of a flash stick. Pick a microdrive that generates as little heat as possible. BTW, what exactly is the motivation for running linux off of a usb stick? If you would like cdr's, you could likely do so with ngrep and a perl script. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Sunday, April 01, 2007 9:08 AM To: Asterisk-Users Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone) Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 31 Mar 2007 16:02:06 -0500 From: Mike Lynchfield [EMAIL PROTECTED] Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works
RE: [asterisk-users] Sponsored development - Monodirectional audio handling
You could put a bounty on this. You may find someone who will be willing to write this for money. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edoardo Serra Sent: Saturday, March 31, 2007 11:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sponsored development - Monodirectional audio handling Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: wireless desktop phones
You can always using a gaming bridge for phones that do not support
wireless.
I've done this before with this:
Linksys / WGA54G / 54Mbps / 802.11g / Wireless Bridge
Setup is pretty easy.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen
Sent: Saturday, March 31, 2007 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: wireless desktop phones
JN == Jordan Novak [EMAIL PROTECTED] writes:
JN Okay, I get it. I still have a problem though. I have no way to
JN wire 30% of these end-points. P{hysically impossible. They do have
JN cat3 twisted pair to each phone. But of course they want IP. Are
JN there any adpaters that will give me just enough bandwidth to get
JN it done. The computer network is all wireless so the phones would
JN have all the bandwidth.
HomePNA should do what you want.
/Benny
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RE: [asterisk-users] One way intermittent static to PSTN
That's a new one. I've seen that on lines when using a card without echo cancellers or LEC T1's. Never with ATT T1's. Make sure you have the newest firmware on your Sangoma card, then try swapping it out. If you have access to a hammer, it might prove worthwhile to hook it up and run a MOS score on the line itself. If it comes up low, then you escalate it with the carrier. Good luck. That's an interesting one. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Porier, Jeremy M. Sent: Friday, March 30, 2007 12:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] One way intermittent static to PSTN We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option 61c and then several IAX trunks running into this box as well. Box is HP DL 385 G2. I've ruled out bad cables, bad port on sangoma and bad port at Level3 rack. When under load and while the problem is occurring zttest is never less than 99.987793 and is usually 100. Nothing showing up in any logs anywhere. Not sure it is related, but I'm noticing a very loud click when an incoming or outgoing call is initiated that I don't remember in the past. I'm stumped. Anyone ever experience this? Suggestions for further trouble shooting? Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] xten web phone
Yeah, that was popular way back. They stopped freely distributing the activex client. If you call Counterpath, they will still license you a copy though. http://www.xten.com (counterpath) -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pezhman Lali Sent: Friday, March 30, 2007 9:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] xten web phone hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Problem converting a Cisco 7960 to SIP
You can go directly to 5.2 and then move on to 7.x and 8.6. 5.2 allows you to upgrade to the newer firmware releases that have an app loader, which Cisco added in later releases. Beware that some cisco non-sip loads can not generate the proper firmware filename to download from tftp when they read the version numbers from the version text. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Wellsted Sent: Thursday, March 29, 2007 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Problem converting a Cisco 7960 to SIP -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brad Stockdale wrote: Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is no VLAN configuration. The Asterisk box also is running a TFTP server and DHCP server. The 4 original Cisco's work fine still. The Polycom IP500's work fine. The problem is with trying to get this new Cisco 7960 online... It came pre-loaded with the SCCP image and I cannot get it to convert to SIP. Currently it is running the following versions: App Load ID: P0030301MFG2 Boot Load ID: PC0303010200 Version: 3.1(MF.G2) The phone contacts the DHCP server and gets an IP successfully. The dhcpd.conf file: To convert a version that old will require you going through at least 3-4 downloads: First convert to SIP 3.3, then to SIP 5.2 then probably to SIP 7.x before finally to SIP 8.6. You may have to convert to a SIP 6.x image in between. After the first conversion, you should be able to set the password etc via the SIPDefault.cfg file. HTH - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRgwFdEtP/KMNOfRbAQIewgf/cJjP9/wvbFp7ibkWgrvKTjY4k4vdhVF2 B5w5INJNjmxSaaC8JsCviglz7+HjFnKoEYd+eIOBYdmzoxGGPNEZLHGqXXd1vqHv EpdxstUZITOdv/gH5TRRzrlzbGWWtyGG8iHC7DftMeBGLyiO1W5/LZDdnpSylD3P NslzlNRQNSPPHjGClBTL3Y1Bsu2MEonICnGoVSyt9g8kHTLpkgeMLJiWWyoKNUcn Uz7Ub1chg8EkV7MgN2veAUKTcLCDIE7kluGC5To22SbcUxwTRnl+IKuE7CgZ+dJN l6zGYyBVdO+hndagOTeBotk2z0xt5D664Fq7NxjkVjUHJTip+nkd9g== =/R9w -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [asterisk-users] SIP RTP Tunnel
You should get a packet capture and look at the SDP that is agreed to by both parties. It sounds like someone is not honoring it. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 29, 2007 9:35 PM To: asterisk-users@lists.digium.com Subject: RE: RE: [asterisk-users] SIP RTP Tunnel Hola Sanjay, this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1. Any idea? Thanx!! -Original Message- From: Sanjay Rajdev [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 29. März 2007 18:27 To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP RTP Tunnel Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). Regards, Sanjay Rajdev - Original Message - From: kalle odenthal [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] SIP RTP Tunnel Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 ---RTP--- Asterisk ---RTP--- SIP2 I know it seems quite useless. But I want to simulate a IAX - SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Question about DSP in Digium card
You've got a decent server. Generally the limiting factor for the number of simultaneous calls is more about server memory. That server could likely handle 124 simultaneous calls, but you would be prudent to double that memory size. Make sure you are running at 100 full especially if you are using G.711. 10 Full uplinks won't cut it if you are running that kind of bandwidth. As for the DSP, you are right to be concerned about the Digium cards, but not because of the DSP. The DSP is not where you will run into problems. Digium cards feature 2 year old circuitry and do not play well with other devices. You have to take care not to share interrupts with any components that may be active on that system. Sharing an IRQ between a Digum card and an Ethernet card would certainly spell disaster in my experience. From personal experience, I no longer use Digium hardware since I could rarely push a quad port card to more than 13 channels per T1 circuit without the card failing miserably. HDLC aborts abound. For now, I only use Sangoma cards. These don't have the IRQ issues and I have had no problems pushing their cards to their maximum. I recommend echo canceller enabled cards for any T1/E1's you may use that are not long distance carrier lines. Good luck, hope this helps with your capacity planning. - SG ## 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Multi-registration ?
Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. If you are using the same account for both line appearances, theoretically it should work on a phone like a Cisco 7960, but it would behave strangely when calls came in. Both line appearances would indicate an inbound call. If you are using two different accounts, there will be no problems at all. Each line appearance would register and could receive calls on either. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http:// http://VoIPSecurityTraining.com VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Tuesday, March 27, 2007 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-registration ? 2. Is possible to do the same with SIP hardphones ? Some hardphones support registering to multiple sip accounts from one phone. (as indeed do some softphones) Is that what you want ? Yes but my question is : Is it possible to register 2 accounts for the same user and hardphone within the same Asterisk server ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Multi-registration ?
Sorry. My mistake. I was thinking of SER. You are quite right. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Tuesday, March 27, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-registration ? On 27/03/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. No it can't - the latest registration 'wins'. To achieve simutaneous ringing of more than one phone (hard or soft), you need a SIP account for each and an entry in the dialplan which rings both. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Anyone having trouble with claling US Domesticon Sellvoip?
I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: Sunday, March 25, 2007 9:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anyone having trouble with claling US Domesticon Sellvoip? I'm not surprised. On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote: Salvatore Giudice wrote: Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Clearly, sellvoip rocks! -stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
