RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion
Sounds to me like this a wonderful in to voice auctions... Rick -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Monday, September 12, 2005 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion Seems to me that the intellectual property (the network + encryption + client) is really the crown jewel here and an Ebay/Skype client is a no brainer. In fact, I can see a scenario where they drop or severly de-ephasize the voice part for their ends - maybe reduced to a Powered by Skype blurb on the client skin (just like you see Powered by AOL search or other claptrap on CNN) The reality will be how well Ebay handles this. They will probably bumble it. Funny thing is, for the $2.6 bil they spent for IP+client list, they chould have designed and implemented their *own* stuff coupled with the mother of all marketing campaigns (think global: tons of Skype users are Asians - any Asian users on the list care to comment about how this is being percieved in Asia?) May You Live In Interesting Times. -Original Message- From: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED] Sent: Monday, September 12, 2005 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion That theory sounds great on paper, but consider some of the factors below. 1) On the web there are no loyalties. Today it is Skype but tomorrow certainly belongs to GoogleTalk, if they can put Encryption. How difficult is that to do? Whose quality is better? Skype or Googletalk? I read that already 10% of the Skype users have migrated to Googletalk and the numbers are soaring. Googletalk is still in Beta. 2)Paying users of payPal are hardly 5% of the 75 Million that they are boasting of. And the numbers are coming down as PayPal has come to be known as a 'PreyPal' of late. Many have stopped using this when the charges have started ripping them off at 6% of the transfer amount for International payments (even between canada and USA) and about 3.5% within USA. 3)What is the percentage of paying users of Skype? 1% - 2%, may be. These customers may already be Ebay's customers. Ebay seems to be going like Time Warner buying AOL because it's user base. There are several similarities in this deal. Skype is cool, it has good stuff in it. How much effort does it take for Google to add these features and make the googlers start using Google Talk - a month or may be two. My 0.02 cents Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, September 12, 2005 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion While ostensibly people see this purchase as Ebay being a me-too VoIP player with GoogleTalk and Microsoft buying, what's it called, Teleo? , I think there's a deeper plan. Let's look at the numbers: -Huge installed base, like 30 to 70 million users (those are the numbers I hear kicked around) -Secure, encrypted P2P client that works great through firewalls -Name brand recognition. Even my mother in law has heard of Skype and she is a techno-illiterate. -Installed base loyalty. People who use Skype, love it. -Oh yeah, it does voice, too. (sarcasm on purpose: voice is only partially the point) So what Ebay could do with this is use the Skype client as an Ebay portal that they control. Why, do you ask, would they want to do this? Several reasons: -Break away from browsers with security flaws and inherent social-engineering tactics like phishing. Nobody controls the WWW. Ebay now controls the Skype network, controls the encryption, everything. Less fraud for Ebay to deal with, assuming they repackage the client into Ebay 2.0 - download now for free! -Paypal integration - back to voice, wouldn't it be nice to pay for your phone bill (a Skype account - regular PSTN) through PayPal? -Repositioning of Ebay as an online solution provider once the Ebay Way is fleshed out, much as Google has moved beyond search into holy crap stuff like video on demand, mapping, etc. In the end, isn't it a good business plan to become an indispensable, comprehensive middleman instead of a one trick pony? Voice is just one aspect of it. -30-70 million users ain't bad. They may be Ebayers - they may not. However, this gives Ebay a huge installed base that they can play with before they bring new products and services to the general market. Skype users are crazy about Skype, and they will download any client that Ebay shoves down their throat, as long as it's free. Later, once the service is worked out and debugged, *then* they will do micropayments for the service (via PayPal, of course) still, 2.6 bil is a crazy price and I agree that they paid too much. I would have loved to be a fly on the wall and see
RE: [Asterisk-Users] yet another Asterisk and VMware question
This works. I've done it on occasion for testing. However, because virtual PCs rarely operate on a real-time clock, mostly emulating these features, you will find that anything that read/writes to disk will suck badly. For example, it is nearly impossible to use the Voicemail features of Asterisk under Vmware, CoLinux or UserMode Linux. Believe me, I've tried! ;) This is one of the main reasons that AstWind has stagnated. The timing granularity of the virtual machines is not acceptable for doing anything IO related. Just since I am curious, what version of VMWare did you use and what kind of box where you running on? I've just moved my * box to a VM on ESX server and didn't play with voicemail until you mentioned it - now Allison's voice cuts in and out. Sounds like I am going to have to go back to the box I was running on previously. My original box is a P3 500 desktop while my VMWare ESX box is a dual P3 1.4GHz HP Proliant server. Rick The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7206 and Sample configs (Newbie)
Are you going to run Call Manager Express on that router or are you going to use it in a different manner? I've had so-so luck making CME talk to * and back for all functions, but it can work. Rick -Original Message- From: Ronnie Tartar [mailto:[EMAIL PROTECTED] Sent: Saturday, August 06, 2005 5:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7206 and Sample configs (Newbie) Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know If I will be able to use the vg248 but would like to. We have a call center that is currently dormant but are considering opening it back up (approx 200 seats). Also looking for ways to cluster or make it highly available. I looked around, not a whole lot of info on this. Thanks in advance. http://www.designatedsystems.com/dynlink.jsp?link_id=63 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Call manager
I've gotten CME to talk to *, but have not used the plain Call Manager. I'd guess you could use a SIP trunk like the wiki talks about to configure call managed to talk to a SIP termination service. Rick -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Thursday, July 28, 2005 1:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Cisco Call manager Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can you caculate with me?
0.29295 is what I get... 930 seconds is 15.5 minutes times .0189 per minute gives me that.. Rick -Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sent: Thursday, July 28, 2005 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Can you caculate with me? before I accuse somebody to overbill I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers:0.7269 or 0.2929 ??? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: VM on * for CME Install - Solved
I found with some more testing that you have to setup a 5 digit number (or something longer than your phone extensions) to make the voicemail work. Now the trick is making the MWI work. Rick -Original Message- From: Lull, Rick Sent: Friday, July 15, 2005 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: VM on * for CME Install Hi folks- I've got to the point of trying to configure voice mail on the * box for the SCCP/CME phones. The phone can call the voicemail number (8500) and I can hear Allison's voice. Attempts to punch in a voicemail box number or password don't seem to register; keypad presses don't seem to be heard by the * box. The CME configuration has the 'dtmf-relay rtp-nte' command set, so I'm not sure why the * console shows nothing. Any ideas? Also, calling the phones on SCCP/CME just gives a timeout, even with the configuration I used from the wiki. The * console shows a SIP Unavailable message, but then I get a busy signal. Anything it might be missing or I should try? Rick *** Rick Lull, CCNA Enterprise Network Analyst Bon Secours Health System 804.627.5006 BSHSI Network Op Center 800.918.0503 Pager 804.317.8586 Cell The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
I believe you need the OS79xx.txt file to be the P003 file and the SIPDefault.cnf file to have the POS3 file name inside. There are some docs on the wiki about it; upgrading the Cisco phones can be tough. Rick From: Walid Azab [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 26, 2005 10:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the fileOS79XX.TXT from the TFP server. In the file I added the version P0S3-06-0-00 - It starts upgrading firmware - Then I get the following message: (Upgrade Failed - Unauthorized) Any ideas? Please find below my conf files. SIP.CONF [300] username=300 type=friend secret=cisco record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid= 300 SIP000CCE351C07.cnf # SIP Configuration Generic File (start) # Line 1 Settings line1_name: 300 ; Line 1 Extension\User ID line1_displayname: 300 ; Line 1 Display Name line1_authname: 300 ; Line 1 Registration Authentication line1_password: cisco ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname: ; Line 2 Display Name line2_authname: UNPROVISIONED ; Line 2 Registration Authentication line2_password: UNPROVISIONED ; Line 2 Registration Password # Line 3 Settings line3_name: ; Line 3 Extension\User ID line3_displayname: ; Line 3 Display Name line3_authname: UNPROVISIONED ; Line 3 Registration Authentication line3_password: UNPROVISIONED ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname: ; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED ; Line 4 Registration Password # Line 5 Settings line5_name: ; Line 5 Extension\User ID line5_displayname: ; Line 5 Display Name line5_authname: UNPROVISIONED ; Line 5 Registration Authentication line5_password: UNPROVISIONED ; Line 5 Registration Password # Line 6 Settings line6_name: ; Line 6 Extension\User ID line6_displayname: ; Line 6 Display Name line6_authname: UNPROVISIONED ; Line 6 Registration Authentication line6_password: UNPROVISIONED ; Line 6 Registration Password # NAT/Firewall Traversal nat_address: voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 # Phone Label (Text desired to be displayed in upper right corner) phone_label: WaZaB-SIP ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST # Phone prompt/password for telnet/console session phone_prompt: Cisco7960 ; Telnet/Console Prompt phone_password: abc ; Telnet/Console Password # SIP Configuration Generic File (stop) SIPDefault.cnf # Image Version image_version: P0S3-06-0-00 # Proxy Server proxy1_address: 10.150.200.165 # Proxy Server Port (default - 5060) proxy1_port:5060 # Emergency Proxy info proxy_emergency: 10.150.200.165 proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: 10.150.200.165 proxy_backup_port: 5060 # Outbound Proxy info outbound_proxy: outbound_proxy_port: 5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5061 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 0 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Enable VAD (0-disable (default), 1-enable) enable_vad: 0 # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: 1 ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 0 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 11 sip_invite_retx: 6
[Asterisk-Users] Cisco Call Manager with Voicemail on Asterisk Problem
I've followed the wiki and can't get the message waiting indicator to light on the phones connected to the CME. The * console gets messages that registration failed - Jul 20 13:27:33 NOTICE[22017]: chan_sip.c:7708 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.0.100' I get one message for each number that is setup on the CME box. Any ideas on what I might be missing? Rick *** Rick Lull, CCNA Enterprise Network Analyst Bon Secours Health System 804.627.5006 BSHSI Network Op Center 800.918.0503 Pager 804.317.8586 Cell The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VPN's
Possibly a gateway problem, on how the phone's IP settings are configured? So far, that is what I have seen with one way audio. Rick -Original Message- From: Peter Osborne [mailto:[EMAIL PROTECTED] Sent: Friday, July 15, 2005 2:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VPN's Hi All, I'm using Asterisk for my PBX, I have a remote office that is connected by a VPN link. I am using Openswan on my side and a Linksys box on the remote side. I have a Polycom IP300 on the remote side configured with a static IP address. When I call the phone on the remote side, it rings and establishes the call fine. The problem I am having is that the remote side can hear the call find but the local side hears nothing. Because of the VPN there are no firwalls in the way. Does anyone have some ideas or atleast how I can track down the problem. Thanks, Pete ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VM on * for CME Install
Hi folks- I've got to the point of trying to configure voice mail on the * box for the SCCP/CME phones. The phone can call the voicemail number (8500) and I can hear Allison's voice. Attempts to punch in a voicemail box number or password don't seem to register; keypad presses don't seem to be heard by the * box. The CME configuration has the 'dtmf-relay rtp-nte' command set, so I'm not sure why the * console shows nothing. Any ideas? Also, calling the phones on SCCP/CME just gives a timeout, even with the configuration I used from the wiki. The * console shows a SIP Unavailable message, but then I get a busy signal. Anything it might be missing or I should try? Rick *** Rick Lull, CCNA Enterprise Network Analyst Bon Secours Health System 804.627.5006 BSHSI Network Op Center 800.918.0503 Pager 804.317.8586 Cell The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco CME Integration - IOS Version known to work?
Hi folks- I'm working on getting a test Call Manager Express system working with Asterisk. My plan is to have * support all the voicemail boxes for the CME/SCCP phones. Right now, I can call from a SIP phone to a SCCP phone and back fine. Calls go from Phone-CME-*-Phone and the reverse. Voicemail works for my SIP phones, but does not work for the SCCP phones. I tried to follow the configuration example on the wiki, but some of the commands aren't there - noticeably the ip2ip redirect command. I'm running c3745-ipvoice-mz.123-15.bin, which is 12.3(15). Anybody have this working have this same IOS version? Or have a suggestion on what IOS to use? Rick *** Rick Lull, CCNA Enterprise Network Analyst Bon Secours Health System 804.627.5006 BSHSI Network Op Center 800.918.0503 Pager 804.317.8586 Cell The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
