Re: [asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS
On 28/02/2012 7:58 PM, DHAVAL INDRODIYA wrote: Hi All, I have one question that if my device is registered over TLS on asterisk . is it required that it can only use SRTP for making an outbound calls or incoming calls too. No. how we can disable srtp and only enable TLS. tlsenable=yes Make sure you have your certificates setup prior to enabling this. AFAIK Encryption of RTP isn't on by default, to disable it in your peer configurations use encryption=no is there any dial-plan functions that can help to disable/enable this SRTP. Not that I am aware of. I want following settings. TLS UDP USERAGENT === ASTERISK = VoIpProvider TLS is used for SIP signalling, what you do with RTP is up to you. I have phones configured to use TLS and I have enforced RTP encryption i.e. SRTP by using encryption=yes in the peers configuration an setting the corresponding setting on the phone. When an outgoing call is made to my ITSP the communications between the phone and Asterisk is all encrypted, the communications with my ITSP are all un-encrypted. I have also used encryption=allow, this permits the administrator of the UA to decide if it should use SRTP or otherwise traditional RTP is used. So is it possible with asterisk. Yes! Was that one question!? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] better timing source for an asterisk gateway
Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isdn). I have a TE220BF available than i can use for dahdi timing source. Is a good idea, or this will give me zero benefit for timerfd timing source (will host this gateway on debian squeeze or centos 6.2) ? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk auto-dial out a SIP .
On Tuesday 28 February 2012, upendra wrote: hi, an anyone tell me how to do auto dial to a SIP in Asterisk using a script .any example will be more helpful ... ! Regards Upendra You need to inject a callfile into /var/spool/asterisk/outgoing . The file should look something like this (representing a call from extension 301 -- here the technology is important -- to extension 101 in context internal); 8 Channel: SIP/301 Context: internal Extension: 101 Priority: 1 CallerId: 301 8 It's easy enough to start with a complete callfile with placeholders, in a scalar variable; then have your script do regular expression substitutions to fill in the values. You must also be careful that Asterisk will not try to parse the callfile while it is in an incomplete state. The canonical method to do this is to create it somewhere else on the same physical filesystem and `mv` it to the desired location. In practice, if the callfile is smaller than one block, then you probably will get away with just creating it in situ. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alphanumeric DTMF !?
Just for fun I did something similar at one point. 0-9 A-D and * and # make a character set of 16 characters, perfect for encoding as hex. Take your string, get the ASCII value of each character, convert it to hex, and add it to the encoded string. Just before dialing, replace all e with # and all f with * in your encoded string Make the call to the remote system and send the digits in the encoded string, you will need something on the other end to decode the text. Took about 30 seconds to send Hello World! because of limitations in Asterisk's maximum digits on Read (about 40 digits) and needing to ACK packets of 32 characters each and timeouts, etc. I think I used CRC8 to validate the received packets. Overall it was a cool hack and totally impractical in the real world. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Tuesday, February 28, 2012 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Alphanumeric DTMF !? Yeah I know about A-D but can we send more than those !? I've read about h245-alphanum thing but that is definitely not in asterisk, so what other options are there is I've to send more than just A-D ? On Tue, Feb 28, 2012 at 12:42 PM, Matt Darnell mattdarn...@gmail.com wrote: On Mon, Feb 27, 2012 at 8:23 PM, Sammy Govind govoi...@gmail.com wrote: Hi list, What possibilities are there in asterisk to send an alphanumeric DTMF from/to asterisk !? Regards, Sammy Do you mean A-D? You send those just like 0-9*# -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] better timing source for an asterisk gateway
On 02/28/2012 09:06 AM, ml asterisk wrote: Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isdn). I have a TE220BF available than i can use for dahdi timing source. Is a good idea, or this will give me zero benefit for timerfd timing source (will host this gateway on debian squeeze or centos 6.2) ? It is doubtful that using a hardware timing source is going to make any appreciable difference over using res_timing_timerfd or res_timing_dahdi (with the DAHDI core software timer). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The message does not contain any threats AVG for MS Exchange Server (2012.0.1913 - 2114/4837)-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
My first two guesses are that encryption is hosing you or that the single-channel nature of IAX2 may have something to do with it. IAX2 talks on 1 channel, SIP uses twisted pair connotation on two channels (as I understand it). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 3:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?) On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote callers) fixed the call quality issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend usernamesecretcontext=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The message does not contain any threats AVG for MS Exchange Server (2012.0.1913 - 2114/4837) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
encryption=yes is meaningless if the provider doesn't support it (mine doesn't). I put it there in the wild hope they eventually will - and no config change will be needed on my part. Still, when I changed it to encryption=no, and tested there wasn't any difference. So I've tried disabling the jitterbuffer, and encryption, and there's no effect on call quality - outgoing (from me - provider) sounds bad/distorted, while incoming sounds great. On 2012-02-28 21:14:55 +, Danny Nicholas said: My first two guesses are that encryption is hosing you or that the single-channel nature of IAX2 may have something to do with it. IAX2 talks on 1 channel, SIP uses twisted pair connotation on two channels (as I understand it). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Troy Telford Sent: Tuesday, February 28, 2012 3:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds terrible By terrible, I mean skips, stutters, and distortion. It can be difficult (sometimes impossible) to understand. It doesn't matter what codec I use (at least between G.729, GSM, or ulaw). On the other hand: With SIP: - Incoming Voice from my Provider - Asterisk = Sounds great - Outgoing Voice from Asterisk - my Provider = Sounds great The obvious conclusion is to simply use SIP; however as I've said, I'd prefer to use IAX2 - plus, I'm curious why SIP sounds great, while IAX2 only sounds good one-way (ie. incoming to my asterisk system). The server for my provider is identical in either case. So I figure it's one of a few things: - misconfiguration - My ISP (Comcast) is throttling or giving a low priority to IAX, but not SIP - If there's something I can do here, I'd like to know, but I doubt it. - a problem with my provider - In which I'll contact them. For the first case - misconfiguration, I'd appreciate some input. My iax.conf is fairly straightforward: [general] bandwidth=low jitterbuffer=yes forcejitterbuffer=no encryption = yes autokill=yes maxcallnumbers=12 maxcallnumbers_nonvalidated=4 [guest] type=user context=default callerid=Guest IAX User [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes Firewall: Asterisk is behind a connection-tracking firewall; in my case, I've noticed that my own connection to my provider has always been sufficient to allow connection tracking to just work - and incoming calls are accepted without problems, and voice travels in both directions (albeit not so well when outgoing). I have configured my firewall to forward incoming connections on port 4569 to my Asterisk box, and tested. This had no effect on call quality (which is no surprise given it's the /outgoing/ voice that's problematic). Outgoing connections are fairly typical for a NAT setup - anything can go out. Any other ideas before I give up on using IAX? Thanks -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 22:17:37 +, Danny Nicholas said: Ok Steve, obviously you've outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won't go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? I understood Steve to mean the following: - Secure locations like IAX. There's only one port to monitor or allow through a firewall, which is pretty compelling. - Aforementioned locations can't get IAX to work well. - So they hire Steve to get IAX to work properly, and he makes money. At least, that's my take. -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro *Sent:* Tuesday, February 28, 2012 3:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great ** ** PSS ** ** http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). ** ** Forget IAX2 and save yourself time you will never get back. ** ** IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. ** ** Thanks for IAX2 Digium! ** ** Thanks, Steve Totaro ** ** On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?)** ** On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that couldn't hear (in my case, the * box with the DAHDI lines, as the DAHDI callers couldn't hear the remote
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, February 28, 2012 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great PSS http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). Forget IAX2 and save yourself time you will never get back. IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. Thanks for IAX2 Digium! Thanks, Steve Totaro On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?) On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Roger That, I am an IC. I contract with the Government to little ten phone shops. From VA/MD/DC area, I have been contracted and flown in to many large call center locations that were CONUS and OCONUS. My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but my resume speaks the truth. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:44 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 22:17:37 +, Danny Nicholas said: Ok Steve, obviously you've outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won't go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? I understood Steve to mean the following: - Secure locations like IAX. There's only one port to monitor or allow through a firewall, which is pretty compelling. - Aforementioned locations can't get IAX to work well. - So they hire Steve to get IAX to work properly, and he makes money. At least, that's my take. -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
And the dude arrives talking about penis.. On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez car...@televolve.comwrote: I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting information like this, I think I know why but I don't care. You have to have thick skin on these lists, the more technical, the more you better have done your homework or get flamed. It is from years of experience, not outsmarting anyone. It took me months to figure out that it just doesn't work well and as you can see, all of the posts are very dated. Nobody outsmarted anyone, just pure experience and experience of MANY other people that use Asterisk. Many did not wish to make waves and emailed me directly that they either came to the same conclusion or that they switched due to my suggesting and had no more problems. Digium and Digium FanBoys will argue that IAX2 is the best thing since sliced bread. Digium will ALWAYS tow the party line. It was either Flemming or Lesher that actually posted that it was in an official release so it couldn't have bugs. That was the end of listening to Digium about IAX2. That statement was archived with my reply of how ridiculous the statement was. It is all on the mailing list. The compensation thing is very true, people drink the cool-aide about IAX2 and it sounds great. Then it turns out that they go to production, and audio sucks, customers are complaining. It becomes a huge problem obviously to an ITSP or any call center. As I said, my experience is dated, but I have been one of the most prolific people in the Asterisk community, I spoke at Astricon in 2007 on Large Call Center Track and was the #1 poster for the year, a year or two ago. I predate most of Digium Staff. I do this stuff in the real world, over VSAT or whatever connectivity you can think of, my experience is real, not a developer in the world of code. To answer your question, maybe you can spend time and get it to work correctly, I have no idea, but why? Why not just use SIP and be done with it. Also realize that the dated posts have replies that are ridiculous like VoicePulse is probably laying people off right now as we speak. If a challenge drives you and you have tons of time to possibly never figure it out and go to SIP, then by all means, do it. If you want it to just work, use OpenVPN to get your single port, don't believe the Digium party line and replies about using OpenSER or whatever it is called now. I get past the firewall and NAT issues with OpenVPN. My standard now is Vyatta with NTOP, Asterisk, Webmin installed. I only use SIP and use OpenVPN. I build Asterisk from source and menuconfig, I remove all that is not needed, including IAX2. I do download all the sound files in different languages and codecs. It just works. I like things that just work. Thanks, Steve Totaro On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote: Ok Steve, obviously you’ve outsmarted at least this poster. On the one hand, IAX2 has purchased things for you (won’t go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you saying it can be a good thing if you have the time and skill level to pursue it, but beginners should leave it alone? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, February 28, 2012 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great PSS http://bit.ly/ywiwzt On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Google or click this link http://bit.ly/ywiwzteve Steve Totaro IAX and then stop wasting your time, go with SIP even if you need to create VPN tunnel(s). Forget IAX2 and save yourself time you will never get back. IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS, prime contractors to ITSPs around the world. Thanks for IAX2 Digium! Thanks, Steve Totaro On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com wrote: I've tried turning jitterbuffer
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
BTW, Trunking was the other selling point of IAX2 besides using 1 port which is easily a DDOS target and also probably still an implantation problem of using one thread and one proc for all calls. Trunking allowed for less overhead then SIP since all the overhead for the concurrent calls were combined into one stream. Without trunking, you only have the single port thing. It is quite easy to open the correct ports for SIP, some just have GUI with a SIP checkbox, IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. Thanks, Steve T On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov abalas...@evaristesys.comwrote: IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Wow Wikipedia was the only place that had the original meaning and not the slur or slang meaning. A *ghetto* is a section of a city predominantly occupied by a group who live there, especially because of social, economic, or legal issues. The term was originally used in Venice http://en.wikipedia.org/wiki/Venice to describe the area where Jews http://en.wikipedia.org/wiki/Jews were compelled to live. On Tue, Feb 28, 2012 at 6:55 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. Thanks, Steve T On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov abalas...@evaristesys.comwrote: IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Oops, I meant da Asterisk 'hood. Thanks for the protip. On 02/28/2012 06:55 PM, Steve Totaro wrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the word Ghetto means anything above, or a racial slur should look up the true definition. It isn't even an insult to the Asterisk Community. By definition, the Asterisk Community is an online Ghetto. Just wanted to clear that up before someone tries to label you. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 23:29:53 +, Steve Totaro said: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Probably self-torture, yes. I want to at least try to use IAX2 because I can - learn more about Asterisk for the experience, etc. Now that I've found problems, I want to know if it's a problem with my configuration, or if it was my ISP, or perhaps my provider. I have no problem at all with SIP. It seems to be the direction everybody is going - including Digium: From the Asterisk 10 Codecs and Audio Formats page: Note that the additional codecs discussed here are available for use in Asterisk's SIP channel driver, only. Asterisk 10 does not make them available for IAX2, MGCP, SSCP, H.323, UniSTIM, etc. Digium's fax driver doesn't work with IAX2... even in ulaw passthrough mode. So if I can find that yes, it's so much a problem with my configuration but a bug in the software, then I'll be satisfied switch to what works (ie. SIP). -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. For the record: 1.8.8.2~dfsg-1 (via Debian packages). I've tried trunk=no, and it might have made a difference (I'll have a better idea after some more testing.) -- Troy Telford -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:19 PM, Carlos Alvarez car...@televolve.com wrote: Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. Perhaps you should read again and point out where I state that is a router/NAT problem. I said that the configuration of routers and redirecting ports is a pain in the ass for users and creates a lot of support problems that simply don; t exist with IAX2. With a IAX2 ATA you just plug it and works. This cannot be done with SIP and off the shelf cheap ATAs, period. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. -- Alejandro Imass On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my remarks derogatory, I don't particularly care. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 10.2.0-rc2: permitted contact can't register.
An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.0.0 I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. [...] Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my Top posting seems to be more popular due to use broken smart phone MUAs that can't reply in-line. But if you have the means it should be avoided for future reference and direct people to read the archives and find useful information. remarks derogatory, I don't particularly care. You should care. Words like ghetto, your users, and using name brands like Yugo in a pejorative way are all derogatory and may direct the discussions on a personal level which should always be avoided. I don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. We are all here to share our knowledge and our valuable time so to make it worthwhile we should all care about conserving netiquette. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Without trunking, you only have the single port thing. It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT And it is easier for NAT because it uses one port as I stated, next open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true for you but it's certainly not the truth. - SIP requires redirection of ports if behind a NAT which is about 99% of home users, whether behind a WiFi router or an ISP private network. Um, not when the server is on a public IP and the phones are configured correctly. - SIP requires far more set-up and support effort and it's not a valid choice for a simple to use home-phone. (a) ISP routers change IPs frequently, (b) the router may change the ATA's private IP rendering the port redirection broken. What about Magic Jack or Vonage? The phone registers regularly with the server so that negates everything above. I don't do simple home setups, but they are simple home setups, your words, not mine. I have only had to redirect ports if the server is behind a NAT. Get a SNOM 370, flash with OpenVPN, run as a client and no problems, not that there would be anyway. I have placed 20 business phones behind NAT with no special configuration and no issues but a bad phone or two in two years I have hostage negotiators with OpenVPN and a softphone on their laptops, they travel the world and never have problems except maybe bandwidth. - A public SIP (w/o a VPN) requires careful control (e.g. contactpermit in Asterisk) to limit the IPs that can connect to the public box. Else you will get serous harm from things like SIPVicious attacks. This can easily be mitigated by running on nonstandard ports. Fail2Ban, and a ton of other products can help, but yes, you are correct. A competent Admin is required to check logs daily and configure things correctly. ISP change their IPs frequently so maintaining your user/ip list is almost impossible. I use IP=dynamic with no problems but people tying to guess a password that is the extension and MAC of the phone. Dictionary attack is nothing. With a Gig pipe and fail2ban, no problems. Also, I don't know where you live but I got Comcast@home when it first came out and my IP has never changed. ISPs in this area say dynamic but they are static, at least the big two, Verizon and Comcast for home use. IAX2 was very vulnerable as well up to 2009 but many things in this regard have changed and are much better. Granted, these security issues are common for both SIP and IAX2 but IMHO it's easier to manage with IAX. Security was never really the issue if you read the thread. It is about voice quality. - In a NAT scenario SIP requires a couple of redirected ports per extension, which is a no-go for SMB installations requiring several ATAs without going to the extent of installing a more powerful equipment than a simple ATA. Not in my experience, phone registers with server on public IP, no problems except some obscure setting on a firewall. Easy enough to google away. - You may use OpenVPN with SIP as you said but requires a PC which is not an option for a simple VoIP business that delivers something like Vonage, just plug it and it works. Wrong, the SNOM 370 works great with OpenVPN. You just contradicted yourself as far as plug and play. The SNOM 370 can also act as a bridge over the VPN tunnel using the LAN port so the whole office is behind either split tunnel or direct VPN. Any other SIP phone behind the SNOM with VPN bridging will also be on the VPN as well as workstations. AFAIK there is no port redirection or any special configuration to use Vonage and it works almost on any network set-up (I don't use Vonage but know people that do). So if something like Vonage is using SIP it's probably using a VPN software like you recommend. Magic Jack is pure SIP, no VPN Anyway, the point is that SIP and IAX2 have both pros and cons and I don't consider IAX2 to be a broken bat like you state. On the contrary, I think it works pretty well, and we use both SIP and IAX2 targeted to simple Home, SOHO and SMBs that just want to plug it and work. We get that with IAX2 and not with SIP so from our experience is completely the opposite of what you say. That is fine, I added disclaimers and small shops. I deal in the 15,000 calls a day minimum realm, so we live in different worlds. Two cups and and a string work too -- Alejandro Imass IAX2 is supported on cheap ATAs by several chineese companies and they work quite well. IPTables is simple and there are tons of howtos. Thanks, Steve T On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: They said the same thing in 2005, 2008, now
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote: works. This cannot be done with SIP and off the shelf cheap ATAs, period. We do it, so cannot seems to be a strong word. It's not perfect, but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly varies. Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. Also, respect netiquette and don't top post and use derogatory remarks and keep your discussion technical. Unfortunately, top-posting has become normal on this list. I'm just fitting into how others were quoting the conversation. If you find my remarks derogatory, I don't particularly care. I follow the direction of the conversation. If people are top posting, then I follow suit, bottom, then I bottom post, inline as we are, then that is what I do. When in Rome -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Roger That! On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.comwrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
Just to stir the pot a bit, I am a member of a worldwide private network of Asterisk and AstLinux users. the network uses IAX exclusively, and we have no issues relating to audio quality with a large variety of providers, routers, host machines, and expertise in configuration of the specific nodes Many ( in the US and Canada ) use a PSTN connection as well as the private network using voip.ms with equally stellar quality using IAX IAX was chosen as the default network protocol because of the many issues with SIP, routers, and ( later ) the many attempts at break-ins. As an aside, didn't the manufacture of the Yugo die with the death of Yugoslavia? Most of the Yugo's shortly thereafter? If anyone has one now it may be close to the value of a Delorian No replies necessary or even desired. John Novack Carlos Alvarez wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imassa...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. My own home configuration is an Airport Extreme with zero configuration. So either these are very old routers you're having a problem with, or buggy SIP devices, or something else. You should care. Hmm, let me check the reading... http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png don't drive a Yugo but if I did I could easily be offended by the pejorative use of the brand. It's a piece of junk and everyone knows it, including the owners, so who cares? -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. The other option is that the SIP ATA has WAN and LAN ports and the SIP device is being assigned a public IP. SIP does not NAT by itself because it can't, because there is no routing info, it's simply impossible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outbound fax over t38 gateway can't pass
Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500- asterisk- ooh323- cisco 3845- fax machine fax can't pass. always reproducable. as I see in tcpdump produced dump fax machines tries to connect on 9600 and failed, no attempt to down speed. If I send fax in path panasonic 500 extension fax machine - asterisk (ReceiveFAX) it is received successefully all the time. If I send fax from asterisk with SendFax as following: asterisk(SendFax) - panasonic500-asterisk- ooh323- cisco 3845...- fax machine it always passes. Usually on 7200, sometimes on 4800. So ooh323 works OK, fax part works OK, t38 works OK, but not with fax machine (we tested to different). Inbound faxes in reverse direction, i.e. fax machine...cisco3845- ooh323 - asterisk - panasonic - fax machine always pass on 7200. More info is here https://issues.asterisk.org/jira/browse/ASTERISK-19436 Bug report was closed because not a bug :-) Could you help me solve this problem? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Tuesday, February 28, 2012 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.com wrote: On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote: Please expand as to how you set-up a SIP ATA behind a common home router set-up, without port redirection and/or use of a SIP proxy and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN support) it _cannot_ be done. Go buy a WRT-54G or nearly any consumer-class router and just plug in a SIP device. Done. It works. We *never* work on customer routers and very rarely have to tell them to reconfigure their router at all. What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. The other option is that the SIP ATA has WAN and LAN ports and the SIP device is being assigned a public IP. SIP does not NAT by itself because it can't, because there is no routing info, it's simply impossible. _ You (or more correctly your Asterisk or SIP/RTP proxy) will handle all of that NAT stuff for you. The only time I've ever had issues with SIP and NAT is when the router tries to do SIP ALG while Asterisk is also trying to do NAT fixups. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alphanumeric DTMF !?
Eric thats really a nice idea to communicate between two or more of our servers. Make the call to the remote system and send the digits in the encoded string, you will need something on the other end to decode the text. But the other end is not our's but could be any solution which requires to feed alphanumeric DTMF that something on the other end could be a propriety solution like CISCO as I mentioned about its alphanumeric relay. SO, I can't ask the other end to change. I'm having a strong feeling that I shouldn't push further into this as Asterisk has its DTMF methods defined and those don't send Alpha-numeric. that's it - end of line. :( On Tue, Feb 28, 2012 at 9:27 PM, Eric Wieling ewiel...@nyigc.com wrote: Just for fun I did something similar at one point. 0-9 A-D and * and # make a character set of 16 characters, perfect for encoding as hex. Take your string, get the ASCII value of each character, convert it to hex, and add it to the encoded string. Just before dialing, replace all e with # and all f with * in your encoded string Make the call to the remote system and send the digits in the encoded string, you will need something on the other end to decode the text. Took about 30 seconds to send Hello World! because of limitations in Asterisk's maximum digits on Read (about 40 digits) and needing to ACK packets of 32 characters each and timeouts, etc. I think I used CRC8 to validate the received packets. Overall it was a cool hack and totally impractical in the real world. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind Sent: Tuesday, February 28, 2012 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Alphanumeric DTMF !? Yeah I know about A-D but can we send more than those !? I've read about h245-alphanum thing but that is definitely not in asterisk, so what other options are there is I've to send more than just A-D ? On Tue, Feb 28, 2012 at 12:42 PM, Matt Darnell mattdarn...@gmail.com wrote: On Mon, Feb 27, 2012 at 8:23 PM, Sammy Govind govoi...@gmail.com wrote: Hi list, What possibilities are there in asterisk to send an alphanumeric DTMF from/to asterisk !? Regards, Sammy Do you mean A-D? You send those just like 0-9*# -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound fax over t38 gateway can't pass
btw, played with res_fax.conf if I set maxrate=7200 fax machines try (and fail) 9600 anyway. Why? If limited ti 7200? looks like bug... So I set maxrate=4800 and modems=v27. Faxes pass Looks like problems with V29... 29.02.2012 07:56, Dmitry Melekhov пишет: Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500- asterisk- ooh323- cisco 3845- fax machine fax can't pass. always reproducable. as I see in tcpdump produced dump fax machines tries to connect on 9600 and failed, no attempt to down speed. If I send fax in path panasonic 500 extension fax machine - asterisk (ReceiveFAX) it is received successefully all the time. If I send fax from asterisk with SendFax as following: asterisk(SendFax) - panasonic500-asterisk- ooh323- cisco 3845...- fax machine it always passes. Usually on 7200, sometimes on 4800. So ooh323 works OK, fax part works OK, t38 works OK, but not with fax machine (we tested to different). Inbound faxes in reverse direction, i.e. fax machine...cisco3845- ooh323 - asterisk - panasonic - fax machine always pass on 7200. More info is here https://issues.asterisk.org/jira/browse/ASTERISK-19436 Bug report was closed because not a bug :-) Could you help me solve this problem? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.2.0-rc2: permitted contact can't register.
You want to allow single IP or whole subnet ? Regards, Zohair Raza On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com wrote: An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.**0.0 http://69.0.0.0/255.0.0.0 I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
