Re: xfone 018 phone service and Linux
Hi, I have signed today with xfone 018 for the internet service (over ADSL), and found that their website (http://www.018.co.il) is broken regarding internet standarts. Firefox users can not login into the My Account area, and users can not send feedback using the Contact Us feedback box. Just a warning for those who consider them. I am currently in talks with them on the issue. Should anything change, I will post additional message. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
--- On Wed, 5/6/09, Arie Skliarouk sklia...@gmail.com wrote: Eventually I did not went with 018, as they have the ugliest support I ever seen - they don't reply on emails. I asked twice from them to send oraat keva blank, and got nothing. The broken שלך button on צור קשר page (under firefox) did not help either: http://www.018.co.il/contactus.asp I would say that is a general trend in Israel. I have more examples. Either no email address is published, or that you can't make a progress solely by email. Now I am looking in direction of 012 and have couple of questions to actual users of theirs voip service with Linux (if there are any): http://www.012.net.il/sales.aspx?docID=8639FolderID=1005lang=hetabn=0 Does the line management site (http://072web.net) works from firefox? The interface has 2 parts: one that control the more static settings, and the other control the more RT ones. Not sure about the former. The later doesn't. Note the static and RT are from an ordinary user perspective. Do they provide SIP credentials, or are these hidden in the hw sip adapter they provide? Even though I don't know what SIP credentials are, I am pretty sure they are hidden. Can Asterisk be used with the service? Probably not, unless you are looking to connect Asterisk in the same way that you would connect it to a Bezeq line. Thank you in advance! Rumor is that 074 now offers 3 months, no commitment, free ISP + land phone services. -- Arie On Sun, Mar 8, 2009 at 03:01, e2xbegqsdyt21hfc e2xbegqsdyt21...@yahoo.com wrote: I don't know the technical details or 018. I do know that 012 provides you with a sip adapter that is plugged between the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified. There is also a rumor that bezeq's new generation network uses sip over copper for the phone services. --- On Sat, 3/7/09, Arie Skliarouk sklia...@gmail.com wrote: From: Arie Skliarouk sklia...@gmail.com Subject: Re: xfone 018 phone service and Linux To: linux-il linux-il@cs.huji.ac.il Date: Saturday, March 7, 2009, 11:21 PM Hi, On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef gi...@codefidence.comwrote: Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. Theory aside, I would like to hear first-hand experiences of people with 018 before I commit for a year of phone service with them. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il -Inline Attachment Follows- ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
Hi, Eventually I did not went with 018, as they have the ugliest support I ever seen - they don't reply on emails. I asked twice from them to send oraat keva blank, and got nothing. The broken שלך button on צור קשר page (under firefox) did not help either: http://www.018.co.il/contactus.asp Now I am looking in direction of 012 and have couple of questions to actual users of theirs voip service with Linux (if there are any): http://www.012.net.il/sales.aspx?docID=8639FolderID=1005lang=hetabn=0 - Does the line management site (http://072web.net) works from firefox? - Do they provide SIP credentials, or are these hidden in the hw sip adapter they provide? - Can Asterisk be used with the service? Thank you in advance! -- Arie On Sun, Mar 8, 2009 at 03:01, e2xbegqsdyt21hfc e2xbegqsdyt21...@yahoo.comwrote: I don't know the technical details or 018. I do know that 012 provides you with a sip adapter that is plugged between the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified. There is also a rumor that bezeq's new generation network uses sip over copper for the phone services. --- On Sat, 3/7/09, Arie Skliarouk sklia...@gmail.com wrote: From: Arie Skliarouk sklia...@gmail.com Subject: Re: xfone 018 phone service and Linux To: linux-il linux-il@cs.huji.ac.il Date: Saturday, March 7, 2009, 11:21 PM Hi, On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef gi...@codefidence.comwrote: Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. Theory aside, I would like to hear first-hand experiences of people with 018 before I commit for a year of phone service with them. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
On May 6, 2009, at 2:23 PM, Arie Skliarouk wrote: Now I am looking in direction of 012 and have couple of questions to actual users of theirs voip service with Linux (if there are any): http://www.012.net.il/sales.aspx?docID=8639FolderID=1005lang=hetabn=0 • Does the line management site (http://072web.net) works from firefox? • Do they provide SIP credentials, or are these hidden in the hw sip adapter they provide? • Can Asterisk be used with the service? Thank you in advance! I think it would be worth your while to give Orange a call. They provide a similar service and from my experience with their cellular phone service their customer support is excelent. Their normal package includes the ISP half of a landline connection, and 600 minutes a month of service to all phones in Israel. If you make lots of calls to cell phones, it's a good deal. The downsides are they only advertise it with their box, they may allow you to provide your own (which could be an asterisk system using SIP) or not. The other disadvantage is the incoming number is an 054 number which I assume is billed at the cellular rate. A phone call to ascertain that and their rates would be worth it, IMHO. The other option is HOT. I used to have a HOT line, it was removed yesterday. It was as good as a BEZEQ line in quality (it even supported FAX). I dropped it because no one was using it and we have a VoIP line. If you make most of your calls to landlines, it's very cheap. It's not VoIP so you would have to put an FXO card in your switch. :-( Has anyone written a SKYPE to Asterisk bridge? SkypeOUT is cheap if you can stand the quality of service (I can't) Geoff. -- geoffrey mendelson N3OWJ/4X1GM Jerusalem Israel geoffreymendel...@gmail.com ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
In fact, there was an open GPL violation issue with them for quite some time. I believe it has been setteled about a year ago or so. Is there a specific source that has more concrete information on the GPL violations that they were accused of and how it was ended? Regards, Dan ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
Arie Skliarouk wrote: Hi, Recently I discovered an VoIP phone service by company xfone: http://www.018.co.il/mpa.asp From what I understood, they provide you with hardware phone that is connected to regular internet line (preferably with them as the ISP). They also provide an PC client for windows that supposedly allows you to call landlines and mobiles phones in Israel and over the world using your phone account (similar to skype). They provide you with an adapter that has RJ45 to your LAN/router and RJ11 to your POTS (phone). The adapter does DHCP and then goes out to some SIP gateway. I asked nicely, so they gave me two concurrent outgoing calls (so I can call via Softphone and the adapter). Can someone confirm my understanding? Is the PC client an regular SIP softphone? I am not sure, but you can download it and see for yourself. I did some wireshark captures of it and it does SIP (according to wireshark). The more interesting thing I do is use my Wifi capable E65 to dial using Fring with their account (this also makes my E65 a Skype terminal). I am still unable to make the Symbian SIP client use them. Also, I cannot accept calls via Fring (but I don't care, since I don't give this number to people). They have a 29.90 NIS account for 500min/month (landline only), call price is 0.079NIS/min for landline and 0.32NIS/min to mobile. They are cheaper than regular SkypeOut for calls to Israel. I considered their other option (9.90 for 24 months for the adapter and the rates as above), but didn't want to commit for 24 months. They have a few problems. I cannot dial 1-800 numbers (even their own 1-800-078-078). This is still a pilot. I am still trying to figure what extra services they have (voice mail, voice to email, call forward etc). You cannot move your Bezeq number. They had some billing problem with the mobile taarifs (it was ~3NIS instead 0.32NIS), but I called them and they fixed it. How do they solve the latency problems inherent to any internet connection? I didn't notice a problem with that. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il -- -- Yaron. ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
2009/3/17 Yaron Zabary ya...@aristo.tau.ac.il: The more interesting thing I do is use my Wifi capable E65 to dial using Fring with their account (this also makes my E65 a Skype terminal). I am still unable to make the Symbian SIP client use them. Also, I cannot accept calls via Fring (but I don't care, since I don't give this number to people). Check your skype privacy settings if you use Fring+Skype, my experience seems to indicate that they reset your privacy to public (i.e. anyone can call/chat/add and generally spam you). I went back to Skype Lite and now the iSkoot client provided by my mobile provider, and the privecy setting reset seems to be gone (after a few weeks). I still have to find a way to call SIP from E71, with any client. I tried Fring, Gizmo and the built-in client. Should try again with the new firmware I got last week. Cheers, --Amos ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. ahem... accept in real life VoIP packets are rarely larger then 250 bytes (most of the time then are not over 100 byes) and that the SIP protocol doesn't actually carry any media (read: voice) information - that is left for RTP. More to the point - VoIP , in general works great. In the (very distant) past been on the technical team of a VoIP start up and trust me - VoIP works great on ADSL lines. Theory aside, I would like to hear first-hand experiences of people with 018 before I commit for a year of phone service with them. That I agree completely - that fact that VoIP works doesn't means that 018 do... :-) Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
e2xbegqsdyt21hfc wrote: I don't know the technical details or 018. I do know that 012 provides you with a sip adapter that is plugged between the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. SIP doesn't actually carry voice, only the control information. Voice is carried in RTP packets and they are normally around 100 bytes. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified. BTW, I have it from very good sources that there exists newer versions of the AudioCodes adapters that actually run on Linux and there is even an up market version that runs Asterisk... :-) Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
On Mon, Mar 09, 2009 at 03:56:20PM +0200, Gilad Ben-Yossef wrote: BTW, I have it from very good sources that there exists newer versions of the AudioCodes adapters that actually run on Linux AudioCodes adapters have run Linux for quite some time. In fact, there was an open GPL violation issue with them for quite some time. I believe it has been setteled about a year ago or so. (Disclaimer: our company's product is somewhat a competeing one to theirs) -- Tzafrir Cohen | tzaf...@jabber.org | VIM is http://tzafrir.org.il || a Mutt's tzaf...@cohens.org.il || best ICQ# 16849754 || friend ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
Hi, On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef gi...@codefidence.comwrote: Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. Theory aside, I would like to hear first-hand experiences of people with 018 before I commit for a year of phone service with them. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
On Mar 7, 2009, at 11:21 PM, Arie Skliarouk wrote: Hi, On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef gi...@codefidence.com wrote: Arie Skliarouk wrote: On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. SIP packets are about 100 bytes. They are even smaller with header compression, but CISCO has a patent on that, so I can't say for certain anyone else uses it. I have a US VoIP provider and they are extremely well connected to Netvision. Average ping times are slightly over 160ms. I used to have Vonage, and from Wednesday afternoon until early Sunday morning, they were unuseable. My current provider is good 24/6.5 (I'm shomer shabbat), but unlike my vonage connection, it was perfectly usable 3pm Friday, 8pm Saturday. I have a 5m/256k cable modem from hot, and a TP-Link router without QOS. Note there are other VoIP providers that do similar things, 012 provides you with a BEZEQ aDSL line and a box, but won't let you use it for Internet, Orange makes you get the line, but provides the ISP service and box. I have heard BBL does too, but have no desire to find out. Actually the cheap deals for outgoing calls currently are the HOT 077 service with 2,000 minutes a month to BEZEQ and 077 lines. It is not VoIP and can be used for a fax, btw. Orange's 139 NIS for combined ISP service and 600 minutes is a good deal if you use it to call cell phones, as HOT is 45ag a minute. Neither is very good if you call mostly outside of Israel. SKYPE has outgoing only service, but it is SKYPE, which is something some people love and others hate (I'm on the hate side) for $6 a month ($13 for a lot of countries). My wife is in Amsterdam for 4 days, and I seriously thought about spending the $6 for one month of an incoming Dutch number via SKYPE, but she had no landline access. Geoff. -- geoffrey mendelson geoffreymendel...@gmail.com ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
I don't know the technical details or 018. I do know that 012 provides you with a sip adapter that is plugged between the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified. There is also a rumor that bezeq's new generation network uses sip over copper for the phone services. --- On Sat, 3/7/09, Arie Skliarouk sklia...@gmail.com wrote: From: Arie Skliarouk sklia...@gmail.com Subject: Re: xfone 018 phone service and Linux To: linux-il linux-il@cs.huji.ac.il Date: Saturday, March 7, 2009, 11:21 PM Hi, On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef gi...@codefidence.comwrote: Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. On a 150kbit upload ADSL upload of a 1500 bytes packet takes about 85ms. To send out the VoIP packet would take another 85ms (on condition that you have a really good QoS). This causes latency of 85-170ms with jitter 85ms. The VoIP speech (SIP protocol) has a lot of skips and is unacceptable. Theory aside, I would like to hear first-hand experiences of people with 018 before I commit for a year of phone service with them. -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
Re: xfone 018 phone service and Linux
Arie Skliarouk wrote: How do they solve the latency problems inherent to any internet connection? The round trip time to a well connected (read: 0% packet loss) server in Israel from an Israeli ISP, where Israel here is defined as connected to the IIX, is under 50ms. The jitter less then 1ms and as jitter buffer translate jitter to latency, you can just take it as 60ms delay with a huge margin. The latency at which most people start finding delay in telephone calls intolerable is around 500ms (depending on codec and other aspects of course), so there is really no problem. Gilad -- Gilad Ben-Yossef Chief Coffee Drinker Codefidence Ltd. The code is free, your time isn't.(TM) Web:http://codefidence.com Email: gi...@codefidence.com Office: +972-8-9316883 ext. 201 Fax:+972-8-9316885 Mobile: +972-52-8260388 The Doctor: Don't worry, Reinette, just a nightmare. Everyone has nightmares. Even monsters from under the bed have nightmares, don't you, monster? Reinette: What do monsters have nightmares about? The Doctor: Me! ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
xfone 018 phone service and Linux
Hi, Recently I discovered an VoIP phone service by company xfone: http://www.018.co.il/mpa.asp From what I understood, they provide you with hardware phone that is connected to regular internet line (preferably with them as the ISP). They also provide an PC client for windows that supposedly allows you to call landlines and mobiles phones in Israel and over the world using your phone account (similar to skype). Can someone confirm my understanding? Is the PC client an regular SIP softphone? How do they solve the latency problems inherent to any internet connection? -- Arie ___ Linux-il mailing list Linux-il@cs.huji.ac.il http://mailman.cs.huji.ac.il/mailman/listinfo/linux-il
