Re: [asterisk-users] Some IAX calls do not disconnect.
On 7 Jul 2009, at 05:05, Steve Totaro wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Obviously Steve and I don't agree about this. There are places where IAX can go that SIP just can't. When Steve says just use SIP, what he is actually recommending is to use SIP/STUN/SDP/RTP/IPSEC to get the same result. (at a 50% bandwidth overhead) i.e. replace a single 100 page RFC with something like 100 RFCs :-) In a big organization where you control the network infrastructure, that is an entirely viable solution, but when you want to get calls through a messy network without having to fill out an infinite number of change requests to the firewall team you should consider IAX. The mess that SIP makes is reflected in the number of bugs and the code size. I'm currently working with a SIP stack that is about 10x the size of the comparable IAX codebase, which matters in some environments. As to the 'everything over a single port' issue, this is no longer such a big deal. (And it is exactly this feature which provides IAX's firewall penetration) Most modern Linuxes support multiple threads reading datagrams from a single datagram socket. The current IAX implementation in Asterisk doesn't support it, but that's an implementation issue, not the protocol itself. Also IAX now supports redirecting the media - which could be used to send it to a separate port on the same box. Various Digium employees have also badmouthed SIP (I think we all have after a bad day at the SDP coalface), so you can't take such remarks too seriously. I overheard a senior Cisco employee saying So you were right all along about IAX to a very senior Digium employee, which also proves nothing much :-) Competition is a good thing - even amongst protocols. T. Tim Panton - Web/VoIP consultant and implementor
Re: [asterisk-users] Some IAX calls do not disconnect.
On Tue, Jul 7, 2009 at 7:43 AM, Tim Pantont...@westhawk.co.uk wrote: On 7 Jul 2009, at 05:05, Steve Totaro wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Obviously Steve and I don't agree about this. There are places where IAX can go that SIP just can't. When Steve says just use SIP, what he is actually recommending is to use SIP/STUN/SDP/RTP/IPSEC to get the same result. (at a 50% bandwidth overhead) i.e. replace a single 100 page RFC with something like 100 RFCs :-) In a big organization where you control the network infrastructure, that is an entirely viable solution, but when you want to get calls through a messy network without having to fill out an infinite number of change requests to the firewall team you should consider IAX. The mess that SIP makes is reflected in the number of bugs and the code size. I'm currently working with a SIP stack that is about 10x the size of the comparable IAX codebase, which matters in some environments. As to the 'everything over a single port' issue, this is no longer such a big deal. (And it is exactly this feature which provides IAX's firewall penetration) Most modern Linuxes support multiple threads reading datagrams from a single datagram socket. The current IAX implementation in Asterisk doesn't support it, but that's an implementation issue, not the protocol itself. Also IAX now supports redirecting the media - which could be used to send it to a separate port on the same box. Various Digium employees have also badmouthed SIP (I think we all have after a bad day at the SDP coalface), so you can't take such remarks too seriously. I overheard a senior Cisco employee saying So you were right all along about IAX to a very senior Digium employee, which also proves nothing much :-)
Re: [asterisk-users] Some IAX calls do not disconnect.
Steve Totaro wrote: On Tue, Jul 7, 2009 at 7:43 AM, Tim Pantont...@westhawk.co.uk wrote: On 7 Jul 2009, at 05:05, Steve Totaro wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Obviously Steve and I don't agree about this. There are places where IAX can go that SIP just can't. When Steve says just use SIP, what he is actually recommending is to use SIP/STUN/SDP/RTP/IPSEC to get the same result. (at a 50% bandwidth overhead) i.e. replace a single 100 page RFC with something like 100 RFCs :-) In a big organization where you control the network infrastructure, that is an entirely viable solution, but when you want to get calls through a messy network without having to fill out an infinite number of change requests to the firewall team you should consider IAX. The mess that SIP makes is reflected in the number of bugs and the code size. I'm currently working with a SIP stack that is about 10x the size of the comparable IAX codebase, which matters in some environments. As to the 'everything over a single port' issue, this is no longer such a big deal. (And it is exactly this feature which provides IAX's firewall penetration) Most modern Linuxes support multiple threads reading datagrams from a single datagram socket. The current IAX implementation in Asterisk doesn't support it, but that's an implementation issue, not the protocol itself. Also IAX now supports redirecting the media - which could be used to send it to a separate port on the same box. Various Digium employees have also badmouthed SIP (I think we all have after a bad day at the SDP coalface), so you can't take such remarks too seriously. I overheard a senior Cisco employee saying So you were right all along about IAX to
Re: [asterisk-users] Some IAX calls do not disconnect.
Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
Just use SIP and solve all your problems. On Mon, Jul 6, 2009 at 5:00 PM, Tim Pantont...@westhawk.co.uk wrote: Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485 �...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
- Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Tue, Jul 7, 2009 at 12:05 AM, Steve Totarostot...@totarotechnologies.com wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. I think a more useful thing to push for or put effort into is making Speex an industry standard codec. Now that would make alot of sense for everybody. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485 �...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some IAX calls do not disconnect.
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. Every day evening I find that there are about 30 calls in B which is not disconnected. This comprise of both calls from B - A as well as B - C. There are no such lingering calls in A or C. Every day I manually disconnect the calls, shown below are two example with first one from B - C and second B - A. a16-in1*CLI soft hangup IAX2/a16-in1-11080 Requested Hangup on channel 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-a16-q1-16420' == Spawn extension (queue, s, 20) exited non-zero on 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-11080' a16-in1*CLI soft hangup IAX2/a16-in1-903 Requested Hangup on channel 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-sangoma-flip-outgoing-16393' == Spawn extension (inbound-calls, outbound, 1) exited non-zero on 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-903' in iax.conf of B the entries are like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes in C the corresponding entry is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I do not know where even to start. Any idea to resolve this would be much appreciated. raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On 3 Jul 2009, at 07:18, Rajkumar S wrote: Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. Every day evening I find that there are about 30 calls in B which is not disconnected. This comprise of both calls from B - A as well as B - C. There are no such lingering calls in A or C. Every day I manually disconnect the calls, shown below are two example with first one from B - C and second B - A. a16-in1*CLI soft hangup IAX2/a16-in1-11080 Requested Hangup on channel 'IAX2/a16-in1-11080' -- Hungup 'IAX2/a16-in1-a16-q1-16420' == Spawn extension (queue, s, 20) exited non-zero on 'IAX2/a16- in1-11080' -- Hungup 'IAX2/a16-in1-11080' a16-in1*CLI soft hangup IAX2/a16-in1-903 Requested Hangup on channel 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-sangoma-flip-outgoing-16393' == Spawn extension (inbound-calls, outbound, 1) exited non-zero on 'IAX2/a16-in1-903' -- Hungup 'IAX2/a16-in1-903' in iax.conf of B the entries are like: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-in1-a16-q1] type=peer host=192.168.79.176 auth=plaintext secret=password username=a16-q1 qualify=yes trunk=yes in C the corresponding entry is: [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw jitterbuffer=no forcejitterbuffer=no [a16-q1] type=user auth=plaintext secret=password context=inbound-calls qualify=yes trunk=yes I do not know where even to start. Any idea to resolve this would be much appreciated. raj I'd try adding transfer=no in the B iax.conf I'm guessing the box in the middle (B) is somehow transferring itself out of the call but retaining a ghost call entry. It would be interesting to know what state those ghost calls are in - iax2 show netstats on the CLI might tell you something interesting. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf I'm guessing the box in the middle (B) is somehow transferring itself out of the call but retaining a ghost call entry. It would be interesting to know what state those ghost calls are in - iax2 show netstats on the CLI might tell you something interesting. Thanks, I will try these two suggestions also and let know the results. raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: iax2 show netstats The show netstats gives: a16-in1*CLI iax2 show netstats LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/a16-in1-1869 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-4071 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-112621000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-124431000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-131071000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-145261000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-146771000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16384 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-sangoma-flip 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16388 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16389 1000 -10-1 -1 0 -1 0 00 0 0 00 0 IAX2/a16-in1-a16-q1-16391 1000 -10-1 -1 0 -1 0 00 0 0 00 0 I have added transfer=no also, watching for it's effect now. raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
