Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been various machines (all connected via 100mb switch) ranging from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 and GnoPhone. I've tried changing the jitterbuffer settings in iax.conf (including turning it off as I've seen some recommendations on the archives) and I've even tried rebuilding zaptel with the various jitter control switches. At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson
On Tue, 2003-10-07 at 11:14, silverflash@bancclub.net wrote:> At this point I have extension 8500 setup to take me to voicemailmain. When > I connect (IAX only - I do not have any Digium cards in the server at all) I^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ ding ding ding, we have a winner. Please try and install one of the psuedo channels such as ztdummy or the rtc channel driver so as to provide you with timing. Your problems are most likely associated with the lack of something throttling your output.> can generaly not tell what is being said at all. I've used sox and a player > and know that the .gsm files are okay.-- Steven Critchfield <critch@basesys.com>
Thought I would just mention that I have a Pentium 150 with 64MB of RAM, asterisk installed, 2 Budgetone 102's and an X100P. No problem with jitter here or anything like that. I don't use mp3 music on hold because I doubt the hardware would cope particularly well. Has anybody got Asterisk running on anything lower spec than this? Michael On Tue, 7 Oct 2003 silverflash@bancclub.net wrote:> Hello, > > I've been playing around with * for quite a while now, and have run into a > problem that I just cannot seem to figure out. > > When using * and any IAX client (I have tested with GnoPhone and both > clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the > connection. > > What I'm running is a P3-1Ghz machine with 512mb ram for a server. The > other end has been various machines (all connected via 100mb switch) ranging > from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 > and GnoPhone. > > I've tried changing the jitterbuffer settings in iax.conf (including turning > it off as I've seen some recommendations on the archives) and I've even > tried rebuilding zaptel with the various jitter control switches. > > At this point I have extension 8500 setup to take me to voicemailmain. When > I connect (IAX only - I do not have any Digium cards in the server at all) I > can generaly not tell what is being said at all. I've used sox and a player > and know that the .gsm files are okay. > > Anybody have any suggestions of what to try? So far this has been > something I've been playing with before I attempt to put it in a production > system, but so far am not having a whole lot of luck. > > I've not been able to try SIP as of yet, as I've not found a softclient and > the application I will be using * for would require this. > > Thanks, > Mike Atkinson > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I'm coming at this thing from an Operational standpoint rather than a development standpoint. Viewing your problem from that angle, I wonder how well your network is performing. Could you have a cable problem that the Asterisk server hasn't reported (Layer 1); or perhaps your * Server is connecting at 100 megabit/half duplex, but the switch is configured (or auto-detected) 100 megabit/full duplex (Layer 2); or perhaps you have a bad port on your switch (Layer 1 or 2); or perhaps there is a mis-typed subnet mask or default gateway somewhere between the systems that hasn't been caught yet (Layer 3). Is your Asterisk server busy doing anything else that's tying up resources? You also menitoned that you haven't yet found a soft-phone. X-Lite (from www.eten.com) works really well on Windows workstations. Michael T Farnworth <mtf@maximasystems.com> wrote the Oct 7, 2003 12:52 PM:> Thought I would just mention that I have a Pentium 150 with 64MB of RAM, > asterisk installed, 2 Budgetone 102's and an X100P. No problem with > jitter here or anything like that. I don't use mp3 music on hold because > I doubt the hardware would cope particularly well. Has anybody got > Asterisk running on anything lower spec than this? > > Michael > > On Tue, 7 Oct 2003 silverflash@bancclub.net wrote: > > > Hello, > > > > I've been playing around with * for quite a while now, and have run into a > > problem that I just cannot seem to figure out. > > > > When using * and any IAX client (I have tested with GnoPhone and both > > clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the > > connection. > > > > What I'm running is a P3-1Ghz machine with 512mb ram for a server. The > > other end has been various machines (all connected via 100mb switch) ranging > > from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 > > and GnoPhone. > > > > I've tried changing the jitterbuffer settings in iax.conf (including turning > > it off as I've seen some recommendations on the archives) and I've even > > tried rebuilding zaptel with the various jitter control switches. > > > > At this point I have extension 8500 setup to take me to voicemailmain. When > > I connect (IAX only - I do not have any Digium cards in the server at all) I > > can generaly not tell what is being said at all. I've used sox and a player > > and know that the .gsm files are okay. > > > > Anybody have any suggestions of what to try? So far this has been > > something I've been playing with before I attempt to put it in a production > > system, but so far am not having a whole lot of luck. > > > > I've not been able to try SIP as of yet, as I've not found a softclient and > > the application I will be using * for would require this. > > > > Thanks, > > Mike Atkinson > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
>I was wondering if anyone else has had this problem. I have >purchused several Cisco 7940 and 7960 phones. Of the 5 phones so >far I have run accross 2 that that give me malformed TFTP and refuse >to upgrade to the latest version of SIP code -- 5.3. In fact some >of the other phones also give malformed packets do this but they >seem to work OK. The ones giving me the problem when looking in >ethereal are misiing part of the filename to get it upgraded to SIP. > >I know why it does not work, but why are these malformed packets >apearing. I also tried to go to a windows based TFTP server >(original was linux TFTP) with the same results. > >Any ideas anyone? > >BabakSome hints which may get you going: http://www.loligo.com/asterisk/Cisco/79xx/upgrading.79xx.phones Re-name the files on your TFTP server to shorter names; I know I used that trick at least a few times in the past. JT
If they are older phones, you may need to upgrade to version 2.3 before upgrading further. Get a copy of the document "How to Convert a Cisco 7869 CallManager Phone to a SIP phone and the Reverse Process" from the Cisco Web site. Paul Mahler pmahler@signate.com phone: 650-207-9855 fax: 877-408-0105 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Babak Pasdar Sent: Wednesday, October 08, 2003 5:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 7940/60 TFTP Problem I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Babak _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users