Re: [Asterisk-Users] AArgh, * and the 7960
Yes, I've read and implemented all the stuff on IAX. It's the local SIP connection and its RTP streams that's the problem. For instance I noted the strange timestamp behaviour from * on local traffic earlier. Iain --On Tuesday, May 18, 2004 1:56 pm -0600 Rich Adamson [EMAIL PROTECTED] wrote: I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. The problem has been discussed multiple times over the last several weeks. To recap, there is two things needed to incure the problem: 1. cisco 7960 phone (it discards packets with uneven timestamps) 2. asterisk had an iax problem that was fixed about a month ago assoicated with uneven timestamps. The distant iax system will need to be upgraded to fairly recent code. See previous posts for more detail. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:42 pm -0500 Nik Martin [EMAIL PROTECTED] wrote: Out of context, this isn't much information. Is your network connection OK? Yes, AFAIK - I'm running all the traffic shaping / prioritisation stuff mentioned on the list Is your broadband provider having troubles? AFAIK - but then it is BT Openworld ;-) Has some upstream hardware changed that you may not be aware of? My call is going through IAXTEL so Digium must know if there's a problem. A test IVR system within IAXTEL would be nice for testing. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:43 pm -0500 brian [EMAIL PROTECTED] wrote: Strange I do 7960 = * = IAX all day long without one jitter or any bad audio. Now if both ends are NOT running the very latest(within the last month or so) CVS-head for example if you have say a 2 month old chan_iax2.c on one end then oh boy you're in for a bad time they need to update. Is the 7960 using SIP? The problem happens with the latest * (cvs co asterisk). I think it's quite likely the local RTP handling that's the problem. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
I have ethereal installed and I'll do a full call trace. The Catch 22 is I
don't have access to access to a source of repeatable (ie recorded) content
accessed through IAX. That would help in producing traces for the ATA and
7960 for comparison. I mainly use IAX for non-critical international
business calls to people who wouldn't want to be * testers.
Iain
--On Tuesday, May 18, 2004 7:22 pm -0600 brian k. west [EMAIL PROTECTED]
wrote:
Lets look at this and FIX the problem instead of hacking it. What you
need to do is install etherreal and capture a call and parse the
timestamp info to see if they are slipping. Because they are perfect
here.
bkw
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 5:07 PM
Subject: Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network
connection
OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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RE: [Asterisk-Users] AArgh, * and the 7960
I have a 7940 running 6.3 SIP firmware and make the following type of calls:- 7940 = * = IAX2 = * = Digium X100P = Nortel CICS Analog FXS Both local and remote asterisk's run CVS-02/24/04 (built about 30mins apart). The IAX2 connection is over a VPN, and both sites are running 1500k/256k ADSL connections, about 75ms ping time between the sites. The only time I notice any problems is if one site has an application flooding its upstream, otherwise audio quality is very good. The odd packet might drop here and there, scrambling a word or two, which I usually attribute to upstream choking. 7940 is running G.729 over 100Mbps LAN to Asterisk, and IAX2 connection is presently running GSM (I've bought a couple of G729 licences for the remote asterisk but am waiting on the keys to install the beta codec). Unfortunately I don't have any spare 7940/60's at present to try out on the remote * box to see how a SIP-IAX2-SIP call would perform. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
snip But as I've mentioned before, this isn't the whole story. There are other repeatable scenarios that still cause problems, and to which some large progressive providers also see as an issue and won't accept termination becuase of it: GW - SIP - * - IAX2 - * - SIP - 79X0 Now, if this scenario has been corrected as well, please accept my apologies for bringing it up. This config, with the absolute latest CVS HEAD, well as of a week or so ago when I last checked, seems to cause issues on the sequencing. I seem to recall comments that there is some work still being done on getting this cross protocol packet sequencing to work properly? I'll have to get Ethereal out again and prove that it is still happening. And why are we blaming Cisco for dropping packets that are mis-sequenced, when we shouldn't be sending them mis-sequenced packets in the first place? Assuming there is a sequence numbering issue (which I don't doubt, I just haven't taken the time to investigate), no one is blaming cisco. Rather, the sequence problem would be another issue. Cisco's problem is that it drops packets (choppy audio) when the rtp timestamps within the rtp pkt are not consistent. Totally unrelated to sequence numbers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
On Thu, 20 May 2004, Iain Stevenson wrote: The Catch 22 is I don't have access to access to a source of repeatable (ie recorded) content accessed through IAX. That would help in producing traces for the ATA and 7960 for comparison. The payload (i.e. audio) of the RTP stream is not relevant, at least in my experience. All the information you need is in the RTP header -- sequence numbers (not a problem, that I've seen) and timestamps. If you have two SIP phones, a FWD account and an IAXtel account, you have all you need to test SIP-IAX2-SIP. From one SIP phone, use your FWD account to call your IAXtel number, and pick up the incoming call on your other SIP phone. To avoid looping issues (multiple hops through your * box), make the source (FWD) end a SIP client defined directly to FWD, the IAXtel end your * box, and hang your destination SIP client off *. Subject to the bandwidth you have available upstream, this should be an adequate test and allow you to capture everything you need. Capture everything in and out of the * box if you can, as this will give the greatest amount of information and good correlation between the IAX2 traffic and the SIP traffic that goes to your SIP destination. Hope this is helpful (and not restating the bleeding obvious)... Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
The Catch 22 is I don't have access to access to a source of repeatable (ie recorded) content accessed through IAX. That would help in producing traces for the ATA and 7960 for comparison. The payload (i.e. audio) of the RTP stream is not relevant, at least in my experience. All the information you need is in the RTP header -- sequence numbers (not a problem, that I've seen) and timestamps. If you have two SIP phones, a FWD account and an IAXtel account, you have all you need to test SIP-IAX2-SIP. From one SIP phone, use your FWD account to call your IAXtel number, and pick up the incoming call on your other SIP phone. To avoid looping issues (multiple hops through your * box), make the source (FWD) end a SIP client defined directly to FWD, the IAXtel end your * box, and hang your destination SIP client off *. Subject to the bandwidth you have available upstream, this should be an adequate test and allow you to capture everything you need. Capture everything in and out of the * box if you can, as this will give the greatest amount of information and good correlation between the IAX2 traffic and the SIP traffic that goes to your SIP destination. I might add to Vic's comments that simply signing up for an FWD IAX account is enough for testing in most cases. They provide a consistent source of audio in the forms of a milliwatt generator, data-time annoucements, and other automated sources of audio to generate the rtp stream. Some of those sources may have other issues, but they are sufficiently stable to observe sequence numbers, timestamps, etc. It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
www.bkw.org/~web/parse.txt That should parse and show ALL lines where the timestamps slip. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Thursday, May 20, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
http://www.bkw.org/~brian/parse.txt Its still early. :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Thursday, May 20, 2004 8:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] AArgh, * and the 7960 www.bkw.org/~web/parse.txt That should parse and show ALL lines where the timestamps slip. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Thursday, May 20, 2004 8:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. I use two 7960s daily with cvs-head out two cisco gateways and in and out via nufone IAX2 without issues. The question is what is diffrent about your setup vs mine? Brain, As you already know, the timestamp problem with cisco phones dropping packets has been associated with at least two channels. There are lots of folks with iax links to external machines that none of us have any control over, and in many cases, the people managing those external machines are not in a position to update their code just because I have a problem with audio on my Cisco phone. I've got the problem with multiple systems (not under my control) and the path of least resistance to continue using * with 7960's is to remove the rtp.c timestamp code. Nufone has updated their code, but many others have not. From what I understand, the Stable code has not yet been fixed for the iax problem and that most certainly was a major bug. Fair number of folks seem to using stable code as well. Although iax capi are getting hammered, the real issue seems to be that no one has opened a high sev TAC case to fix the root problem. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 19 May 2004 03:45, Vic Cross wrote: Where are you ethereal traces so I can look over them. I appreciate that. http://veejoe.com.au/isdnbadcall.gz http://veejoe.com.au/isdnbadcall2.gz I took a look at those as well and it looks a lot like the problem I'm having only mine is periodical and has nothing to do with Cisco or chan_capi. My setup is two asterisk boxes connected via IAX2 with a SIP phone on one box and playtones() on the other. At random intervals, the timestamp isn't updated. Taking IAX2 out of play, the timestamps seems to be just fine. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAq0dY2TEAILET3McRAvt2AKCh1AHED6sSmsfEhsRTQ8DKJMRE+gCfXsyV wRw5X3aHtWzbleycJGx5yRU= =m3yY -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
brian k. west wrote: You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. I use two 7960s daily with cvs-head out two cisco gateways and in and out via nufone IAX2 without issues. The question is what is diffrent about your setup vs mine? That's a good question. Let me describe my setup and when the problem occurs and when it doesn't and maybe it'll help in figuring out what's really going on here. I have something that looks like this: [PSTN Gateway *] -- 100Mb Ethernet -- [Systemix * svr] -- public Internet -- [local * server] -- 100Mb Ethernet -- [Cisco 7960] (All inter-Asterisk communication is done using IAX) In this configuration with the cvs head from 5/9 the Cisco phone works horribly (dropped/repeated? packets) unless I comment out the timestamp check in rtp.c. Stable seems to work pretty well. Now if I change the config so that it looks like this: [PSTN Gateway *] -- 100Mb Ethernet -- [Systemix * svr] -- public Internet -- [Cisco 7960] Things seem to work ok without having to make changes to rtp.c. So I've had a suspicion that it has something to do with delay or jitter that's introduced in the IAX channel that goes over the public Internet. But, as you say, this works fine for you. Could it be the extra * in the middle? That's one difference between our setups. Now the big unknown here is which version of * my PSTN provider is running. I've asked them. I'm hopeful I'll eventually get an answer. And as far as 'fixing' it goes, I would love to. I'm not without the skill. But, while Asterisk is almost unbelievable in its features set, some of the code is damn hard to grok. Some source files have as many as 8000 lines with virtually *no* comments. I don't think I've seen a single function with a preamble describing what it does, or how it works. Look at the .h files in include/asterisk/ I have, and while there is some documentation there, it's limited and does little to explain the actual code. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
From what I understand, the Stable code has not yet been fixed for the iax problem and that most certainly was a major bug. Fair number of folks seem to using stable code as well. I'm going to state this again for those not paying close attention. CVS-HEAD is just as stable if NOT MORE stable than CVS-STABLE. CVS-HEAD is going to be 1.1-RC1 CVS-STABLE is going to be 1.0-RC1 That's right a dual release. These remote ends can update and be backwards compatible but THEY MUST UPDATE otherwise the problem is going to continue. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
From what I understand, the Stable code has not yet been fixed for the iax problem and that most certainly was a major bug. Fair number of folks seem to using stable code as well. I'm going to state this again for those not paying close attention. CVS-HEAD is just as stable if NOT MORE stable than CVS-STABLE. CVS-HEAD is going to be 1.1-RC1 CVS-STABLE is going to be 1.0-RC1 That's right a dual release. These remote ends can update and be backwards compatible but THEY MUST UPDATE otherwise the problem is going to continue. You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. I'm in favor of backing those rtp.c changes out of both Head and Stable and schedule the re-implementation six months down the road. At least there is a much higher probability the majority of * systems in the universe would have been updated to something that includes the iax and other fixes. Who knows, maybe cisco will have fixed the problem by then too. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
The fact is the provider is running broken code. They should fix it. That's the true bottom line. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
brian wrote: You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. Yeah, right. And just how often should my service provider update their Asterisk installation? Daily? Weekly? Frankly, given that there's not even a 1.0 release of Asterisk I'm amazed any service providers are using it. Asterisk is currently a rapidly moving target, as this very issue demonstrates. Once a 1.0 or 1.1 is released you can bet everyone will upgrade. Until then, all of us should probably keep our expectations in check. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
Quoting brian [EMAIL PROTECTED]: You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. But as I've mentioned before, this isn't the whole story. There are other repeatable scenarios that still cause problems, and to which some large progressive providers also see as an issue and won't accept termination becuase of it: GW - SIP - * - IAX2 - * - SIP - 79X0 Now, if this scenario has been corrected as well, please accept my apologies for bringing it up. This config, with the absolute latest CVS HEAD, well as of a week or so ago when I last checked, seems to cause issues on the sequencing. I seem to recall comments that there is some work still being done on getting this cross protocol packet sequencing to work properly? I'll have to get Ethereal out again and prove that it is still happening. And why are we blaming Cisco for dropping packets that are mis-sequenced, when we shouldn't be sending them mis-sequenced packets in the first place? Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. If your * system has iax links to two service providers, three different companies (in production), and other such things, there is literally no way to convince all remote sites to upgrade to dev cvs. That IS the problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
The fact is the provider is running broken code. They should fix it. That's the true bottom line. Agreed but Rich needs a workaround. I think what I suggested will work and is cheap -- a spare PC with a pair of ethernet cards in it, and a second ethernet card for his existing * server. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
I commented the two lines in rtp.c, took 20 seconds, rebuilt the source, and off we go. Fixes the iax issue just fine. I think the rtp.c 'hack' is the way to go. Bad 'ole 'hack'. Just my 2c. - Chris Netlabz, Inc. - Original Message - From: Ray Burkholder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 19, 2004 11:13 AM Subject: RE: [Asterisk-Users] AArgh, * and the 7960 Quoting brian [EMAIL PROTECTED]: You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. But as I've mentioned before, this isn't the whole story. There are other repeatable scenarios that still cause problems, and to which some large progressive providers also see as an issue and won't accept termination becuase of it: GW - SIP - * - IAX2 - * - SIP - 79X0 Now, if this scenario has been corrected as well, please accept my apologies for bringing it up. This config, with the absolute latest CVS HEAD, well as of a week or so ago when I last checked, seems to cause issues on the sequencing. I seem to recall comments that there is some work still being done on getting this cross protocol packet sequencing to work properly? I'll have to get Ethereal out again and prove that it is still happening. And why are we blaming Cisco for dropping packets that are mis-sequenced, when we shouldn't be sending them mis-sequenced packets in the first place? Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
Brian Cuthie [EMAIL PROTECTED] wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. As I understand it, chan_capi has been released under the GPL. That being the case, the author doesn't need to sign over his copyright or release it as no-license public domain code, and the Asterisk maintainers are free to include it in the CVS tree if they feel that it might be useful. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. A TDMoE link between two * boxes will not retime *everything* passing between them?! problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. Actually, and I do not mean to be rude here, but if you're using * in a production environment you are already part of the problem -- * has never claimed to be production ready. I use it in a production environment, Jeremy McNamara does, many people do but if it doesn't work it's not suddenly someone else's problem that beta/alpha software is not production ready. I am positive that a TDMoE link between *1 and *2 will re-time everything passing between them since you are converting from RTP to TDM and back to RTP (in the case of SIP on each end, or even SIP on one and IAX2 on the other, although IAX2 does not use RTP you're still completely retiming everything)... Worth a shot anyway, innit? -A. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
You can't consider CVS_HEAD dev anymore as its been allowed to stabilize in the past month and will be 1.1 Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, May 19, 2004 11:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. If your * system has iax links to two service providers, three different companies (in production), and other such things, there is literally no way to convince all remote sites to upgrade to dev cvs. That IS the problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
On Wed, 2004-05-19 at 10:12, Brian Cuthie wrote: Asterisk is currently a rapidly moving target, as this very issue demonstrates. Once a 1.0 or 1.1 is released you can bet everyone will upgrade. Until then, all of us should probably keep our expectations in check. Actually CVS -head is the rapidly moving target. CVS -stable gets very few changes. That's why I use -stable. I don't want things changing on me every time I upgrade (other than bug fixes). If you want to see the CVS changes to both -stable and -head then sign up for the CVS mailing list at http://lists.digium.com/ I asked why the rtp.c fixes for bad audio were not put in -stable and I got this response: http://lists.digium.com/pipermail/asterisk-users/2004-April/043296.html -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
Yes they MUST disclaim the code as digium has a dual lic. so digium must have permission to add it to CVS that is why no GPL code can touch asterisk. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, May 19, 2004 10:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] AArgh, * and the 7960 Brian Cuthie [EMAIL PROTECTED] wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. As I understand it, chan_capi has been released under the GPL. That being the case, the author doesn't need to sign over his copyright or release it as no-license public domain code, and the Asterisk maintainers are free to include it in the CVS tree if they feel that it might be useful. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
On Wed, 2004-05-19 at 10:38, Kevin Walsh wrote: As I understand it, chan_capi has been released under the GPL. That being the case, the author doesn't need to sign over his copyright or release it as no-license public domain code, and the Asterisk maintainers are free to include it in the CVS tree if they feel that it might be useful. Ah, the innocence of youth. Try these two URLs: http://lists.digium.com/pipermail/asterisk-users/2003-July/016694.html and http://lists.digium.com/pipermail/asterisk-users/2002-October/005407.html This issue has been beaten to death already. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Kevin Walsh wrote: Brian Cuthie [EMAIL PROTECTED] wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. I'm almost certain I didn't say this. Please be careful with your attributions. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
The fact is the provider is running broken code. They should fix it. That's the true bottom line. Agreed but Rich needs a workaround. I think what I suggested will work and is cheap -- a spare PC with a pair of ethernet cards in it, and a second ethernet card for his existing * server. And the workaround is really undoing the rtp.c changes as Chris has pointed out several times. (Don't need another PC.) The argument here is the rtp.c changes to undo the timestamps should really be completed at the cvs level, and not at an individual * level. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
What's the story on the enhanced voicemail features ? When will they be committed ? They don't appear to be in CVS-HEAD-05/19/04. Thanks, Chris Netlabz, Inc. - Original Message - From: brian [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 19, 2004 12:04 PM Subject: RE: [Asterisk-Users] AArgh, * and the 7960 You can't consider CVS_HEAD dev anymore as its been allowed to stabilize in the past month and will be 1.1 Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, May 19, 2004 11:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. If your * system has iax links to two service providers, three different companies (in production), and other such things, there is literally no way to convince all remote sites to upgrade to dev cvs. That IS the problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Andrew Kohlsmith wrote: The fact is the provider is running broken code. They should fix it. That's the true bottom line. Agreed but Rich needs a workaround. I think what I suggested will work and is cheap -- a spare PC with a pair of ethernet cards in it, and a second ethernet card for his existing * server. No... Service providers that adopted Asterisk this early in the game should have been prepared to dedicate resources to solving problems and hell even contribute to the further development of Asterisk. Yet, I don't see this happening from many of the other Asterisk based service providers. IMHO, it shows who's in this just for the money. Just my $0.02 Peso's, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
They are there.. read the configs/voicemail.conf.sample Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Clifton Sent: Wednesday, May 19, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 What's the story on the enhanced voicemail features ? When will they be committed ? They don't appear to be in CVS-HEAD-05/19/04. Thanks, Chris Netlabz, Inc. - Original Message - From: brian [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 19, 2004 12:04 PM Subject: RE: [Asterisk-Users] AArgh, * and the 7960 You can't consider CVS_HEAD dev anymore as its been allowed to stabilize in the past month and will be 1.1 Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, May 19, 2004 11:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. If your * system has iax links to two service providers, three different companies (in production), and other such things, there is literally no way to convince all remote sites to upgrade to dev cvs. That IS the problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
How does this have anything to do with this thread: Re: [Asterisk-Users] AArgh, * and the 7960. Bill Doll Jr Chris Clifton [EMAIL PROTECTED] Chris Clifton [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/19/2004 09:42 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] AArgh, * and the 7960 What's the story on the enhanced voicemail features ? When will they be committed ? They don't appear to be in CVS-HEAD-05/19/04. Thanks, Chris Netlabz, Inc. - Original Message - From: brian [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 19, 2004 12:04 PM Subject: RE: [Asterisk-Users] AArgh, * and the 7960 You can't consider CVS_HEAD dev anymore as its been allowed to stabilize in the past month and will be 1.1 Bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, May 19, 2004 11:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AArgh, * and the 7960 We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. Fine then -- fix your problem -- set up two * boxes, one talking ot the SIP gateway and then set up an IAX-IAX or better yet TDMoE link between the interim one and the one your cisco phones are connected to. I think the point you're trying to make is that Cisco needs to solve their problem, and that Asterisk needs to make it happen. That isn't gonna work. A second box with a pair of ethernet cards in it will, unless I'm missing something. Two * boxes does not fix the problem as the timestamps will trickle through each. The root of the problem is two fold: a) iax conversations where pkts originate from an * system that is either older then about 30 days ago (or a system based on stable code), and, b) cisco 7960 issue with dropping any packet with uneven timestamps. If your * system has iax links to two service providers, three different companies (in production), and other such things, there is literally no way to convince all remote sites to upgrade to dev cvs. That IS the problem, but there seems to be a major issue with a few on this list understanding that and understanding what production means. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users inline: graycol.gifinline: pic19169.gifinline: ecblank.gif
Re: [Asterisk-Users] AArgh, * and the 7960
Andrew Kohlsmith wrote: The fact is the provider is running broken code. They should fix it. That's the true bottom line. I did not write that, Brian did. Agreed but Rich needs a workaround. I think what I suggested will work and is cheap -- a spare PC with a pair of ethernet cards in it, and a second ethernet card for his existing * server. I wrote that. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. Just to thow in my comments here, time and time again we have upgraded all our boxes to the latest CVS Head (and even by doing fresh checkouts). The SIP-IAX2-SIP Timestamp issue has continued (no other Service Providers involved in our tests). We have opened multiple bugs (and we provided that rtp.c hack in one of those bugs). Our opinion is that a system, such as * that converts media streams from one type of channel to another or from one codec to another should NOT carry over the TimeStamp. Instead it should generate a fresh new one on each outgoing stream, that way guaranteeing a consistent delta spacing. We are quite happy with the rtp.c hack and will continue to use it for the time being. Andres We as users of lots of other service providers and systems other then Nufone don't have the choice of forcing those systems to either Head or Stable. That's purely irrelevant. Since we can't force others to upgrade to anything, we're stuck with either throwing away the Cisco phones or removing the rtp.c code that's causing the poor quality issue. I'm in favor of backing those rtp.c changes out of both Head and Stable and schedule the re-implementation six months down the road. At least there is a much higher probability the majority of * systems in the universe would have been updated to something that includes the iax and other fixes. Who knows, maybe cisco will have fixed the problem by then too. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
Out of context, this isn't much information. Is your network connection OK? Is your broadband provider having troubles? Has some upstream hardware changed that you may not be aware of? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Tuesday, May 18, 2004 1:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AArgh, * and the 7960 I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AArgh, * and the 7960
Strange I do 7960 = * = IAX all day long without one jitter or any bad audio. Now if both ends are NOT running the very latest(within the last month or so) CVS-head for example if you have say a 2 month old chan_iax2.c on one end then oh boy you're in for a bad time they need to update. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Tuesday, May 18, 2004 1:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AArgh, * and the 7960 I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. The problem has been discussed multiple times over the last several weeks. To recap, there is two things needed to incure the problem: 1. cisco 7960 phone (it discards packets with uneven timestamps) 2. asterisk had an iax problem that was fixed about a month ago assoicated with uneven timestamps. The distant iax system will need to be upgraded to fairly recent code. See previous posts for more detail. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network connection OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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Re: [Asterisk-Users] AArgh, * and the 7960
Lets look at this and FIX the problem instead of hacking it. What you need
to do is install etherreal and capture a call and parse the timestamp info
to see if they are slipping. Because they are perfect here.
bkw
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 5:07 PM
Subject: Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network connection
OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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Re: [Asterisk-Users] AArgh, * and the 7960
On Tue, 18 May 2004, brian k. west wrote: Lets look at this and FIX the problem instead of hacking it. What you need to do is install etherreal and capture a call and parse the timestamp info to see if they are slipping. Because they are perfect here. bkw I'd love to fix the problem, but no-one is listening! I did what you said, captured Ethereal traces, found that timestamps do not increment, found BLATANT errors in rtp.c where a signed int is being used to hold return values from an unsigned int function... and had my bug report thrown out because I am only able to reproduce the problem with chan_capi. Now I know that chan_capi doesn't belong to Digium, and I know that you're all trying to get a 1.0 release out. But this problem is really hurting my business, and right now destroying any chance that I might start offering Asterisk as part of commercial solutions. Now, kapejod is not replying to my e-mails, and markster's suggestion (from another bug report) of zeroing out the delivery field in chan_capi's read function did not work. So hacking is all I have left if I want to keep using Asterisk -- which I do, because I think it's a great program with a pretty good community around it. Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
I'd love to fix the problem, but no-one is listening! I did what you said, captured Ethereal traces, found that timestamps do not increment, found BLATANT errors in rtp.c where a signed int is being used to hold return values from an unsigned int function... and had my bug report thrown out because I am only able to reproduce the problem with chan_capi. The problem isn't with asterisk chan_capi will have to be updated to deal with the changes. Now I know that chan_capi doesn't belong to Digium, and I know that you're all trying to get a 1.0 release out. But this problem is really hurting my business, and right now destroying any chance that I might start offering Asterisk as part of commercial solutions. I don't see these issues in any other channel driver. Now, kapejod is not replying to my e-mails, and markster's suggestion (from another bug report) of zeroing out the delivery field in chan_capi's read function did not work. So hacking is all I have left if I want to keep using Asterisk -- which I do, because I think it's a great program with a pretty good community around it. Where are you ethereal traces so I can look over them. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. bkw - Original Message - From: brian k. west [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 8:01 PM Subject: Re: [Asterisk-Users] AArgh, * and the 7960 I'd love to fix the problem, but no-one is listening! I did what you said, captured Ethereal traces, found that timestamps do not increment, found BLATANT errors in rtp.c where a signed int is being used to hold return values from an unsigned int function... and had my bug report thrown out because I am only able to reproduce the problem with chan_capi. The problem isn't with asterisk chan_capi will have to be updated to deal with the changes. Now I know that chan_capi doesn't belong to Digium, and I know that you're all trying to get a 1.0 release out. But this problem is really hurting my business, and right now destroying any chance that I might start offering Asterisk as part of commercial solutions. I don't see these issues in any other channel driver. Now, kapejod is not replying to my e-mails, and markster's suggestion (from another bug report) of zeroing out the delivery field in chan_capi's read function did not work. So hacking is all I have left if I want to keep using Asterisk -- which I do, because I think it's a great program with a pretty good community around it. Where are you ethereal traces so I can look over them. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
On Tue, 18 May 2004, brian k. west wrote: The problem isn't with asterisk chan_capi will have to be updated to deal with the changes. Only someone with knowledge of the internals of * would know that the RTP timestamps generated by * on an outgoing SIP leg would be affected by the incoming channel type. I saw a problem with RTP, I logged it. I don't see these issues in any other channel driver. Okay, so now I know better, that the incoming channel does affect the outgoing RTP, I have something better to go on. Was the big RTP change in * (circa mid-March) discussed anywhere? Is there detail about what needs to change in the channel drivers? If someone can point me at some info, I'll make the [EMAIL PROTECTED] change to chan_capi myself. I'll even have a look at chan_sccp since I've played with that in the past. BTW, as I mentioned before it does take more than zeroing the delivery field in the read function, as I tried this without success. Is there a CVS-web of the * tree? I don't know how to drive CVS to give changelogs etc... Again, if there's a way for me to find out how/what to change, I can give it a go. Where are you ethereal traces so I can look over them. I appreciate that. http://veejoe.com.au/isdnbadcall.gz http://veejoe.com.au/isdnbadcall2.gz Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
On Tue, 2004-05-18 at 20:45, Vic Cross wrote: Is there a CVS-web of the * tree? I don't know how to drive CVS to give changelogs etc... Again, if there's a way for me to find out how/what to change, I can give it a go. http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+%22Getting+info+about+changes+in+CVS%22btnG=Search -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. And as far as 'fixing' it goes, I would love to. I'm not without the skill. But, while Asterisk is almost unbelievable in its features set, some of the code is damn hard to grok. Some source files have as many as 8000 lines with virtually *no* comments. I don't think I've seen a single function with a preamble describing what it does, or how it works. And I don't mean any offense by this. As I said, Asterisk is a truly amazing piece of software. But if the original developers, who really know how this stuff works, could put some effort into documenting the code with some comments, their efforts will pay off ten-fold when others are able to start helping them maintain it. Cheers, -brian brian k. west wrote: Also on a side note if Kapejod isn't wanting keep chan_capi up to date then someone needs to ask him if he will disclaim it so digium can include it and help maintain it. bkw - Original Message - From: brian k. west [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 8:01 PM Subject: Re: [Asterisk-Users] AArgh, * and the 7960 I'd love to fix the problem, but no-one is listening! I did what you said, captured Ethereal traces, found that timestamps do not increment, found BLATANT errors in rtp.c where a signed int is being used to hold return values from an unsigned int function... and had my bug report thrown out because I am only able to reproduce the problem with chan_capi. The problem isn't with asterisk chan_capi will have to be updated to deal with the changes. Now I know that chan_capi doesn't belong to Digium, and I know that you're all trying to get a 1.0 release out. But this problem is really hurting my business, and right now destroying any chance that I might start offering Asterisk as part of commercial solutions. I don't see these issues in any other channel driver. Now, kapejod is not replying to my e-mails, and markster's suggestion (from another bug report) of zeroing out the delivery field in chan_capi's read function did not work. So hacking is all I have left if I want to keep using Asterisk -- which I do, because I think it's a great program with a pretty good community around it. Where are you ethereal traces so I can look over them. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AArgh, * and the 7960
You know, I'm not so sure this is limited to chan_capi. I have two asterisk boxes running, with one connected to my PSTN gateway (also using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head works if I comment out the offending lines. Without commenting them out, the cisco phones drop packets like crazy. No chan_api is involved. I use two 7960s daily with cvs-head out two cisco gateways and in and out via nufone IAX2 without issues. The question is what is diffrent about your setup vs mine? And as far as 'fixing' it goes, I would love to. I'm not without the skill. But, while Asterisk is almost unbelievable in its features set, some of the code is damn hard to grok. Some source files have as many as 8000 lines with virtually *no* comments. I don't think I've seen a single function with a preamble describing what it does, or how it works. Look at the .h files in include/asterisk/ bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
