Re: [Asterisk-Users] Asterisk echo & fxotune
Dan Elder wrote: Hey all, guess I didn't include enough info in my last query. I'm having massive echo problems on our sip<=>fxo connections & have been reading up on echo cancellation & such. When I try to run fxotune I get the following error: Could not fill input buffer Tuning module 25.Failure! The system has two of the digium 4port fxo cards & one T1 card (which isn't fully configured yet) (1-24 are the T1card) Has anyone seen this error? the only hits I've found on google all point to the source code. This happens even if I remove the T1 card. Thx in advance for any insight. I've seen it here when I did not have my FXO's connected to the telephone line. JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Rich Adamson wrote: Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. You can probably search the -cvs list to find it, but that might be a little time consuming. You should see the card's pic id's in dmesg and then look in the zaptel src directory for matching entries, or, simply call digium support. It sounds like you are running an older version of zaptel/asterisk. Thanks again Rich for the info. This is all from latest CVS though. I have generated an e-mail support ticket with digium, so I am looking forward to the answer. No doubt it will be too obvious :). JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
James B. MacLean wrote: Rich Adamson wrote: From: "James B. MacLean" <[EMAIL PROTECTED]> Asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 ---End of Original Message- The above does indicate a problem. The Rev E/F card is known to have issues, and most of the issues revolved around unusual failures after a week or so. But there have been several other changes leading up to the Rev I card (the latest is Rev J with only minor changes since Rev I). I don't know of anyone that has attempted to mix to Rev's of the TDM card in a system, so unknown whether that might be an issue or not. I'd contact digium support and have that Rev E/F card rma'ed under warranty. (All TDM cards are still under warranty.) Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. Next I'll try with just one card, but that will be another day as the machine is not local. thanks again, JES Booting with only one card did _not_ work. Tried each separately. Plugged into phone lines and not plugged into phone lines. I had expected at least that my Rev I card should have worked :(. JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Second TDM22B board install issue
Hi Folks, Sent this to support, but thought it may be obvious and I should pass it here too :). First (older) TDM22B is in use and appears fine. Second one, installed and ztfg -vv reports : [4626926.318000] ACPI: PCI Interrupt :01:08.0[A] -> Link [APC1] -> GSI 16 (level, high) -> IRQ 16 [4626926.363000] Freshmaker version: 71 [4626926.363000] Freshmaker passed register test [4626928.073000] Module 0: Installed -- AUTO FXS/DPO [4626929.304000] Module 1: Installed -- AUTO FXS/DPO [4626929.504000] Module 2: Installed -- AUTO FXO (FCC mode) [4626929.704000] Module 3: Installed -- AUTO FXO (FCC mode) [4626929.705000] Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) [4626929.706000] Registered tone zone 0 (United States / North America) [4627086.73] Registered tone zone 0 (United States / North America) [4627088.903000] Registered tone zone 0 (United States / North America) [4627091.331000] Registered tone zone 0 (United States / North America) /etc/zaptel.conf has: fxsks=3-4,7-8 fxoks=1-2,5-6 When ever I try to use the new channels (5,6,7,8) I may or may not (but usually) get : Nov 25 08:51:51 ERROR[27602]: chan_zap.c:10250 setup_zap: Unable to reconfigure channel '5' Nov 25 08:51:51 WARNING[27602]: chan_zap.c:11010 reload: Reload of chan_zap.so is unsuccessful! /etc/asterisk/zapata.conf has rxgain=10.0 txgain=-5.0 signalling=fxs_ks context=outbound group = 6 callerid=asreceived channel => 5 I am hoping that I simply misconfigured something ? Thanks, JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Dial ZAP with group (g2) erroneously says call answered when it is still ringing
Oh boy :(. As Roman politely explained in a private email... I was using ports 1 and 2 thinking they were the outbound fxs ports :(. That's it, these glasses are going, and no more testing from home :). When I switched to testing with ports 3 and 4, everything worked the same as G2. Not of course as cute as what I had hoped for when I see the local telco can do something like "Dial(ZAP/g2/&SIP/[EMAIL PROTECTED])" and have it wait 'til the correct phone is answered :(. Thanks to C F for the "c" option but my goal was to just have the 4 digit number call folks with and without SIP. I would not expect users to know to press #. I don't think dvlinedetect will quite cut it either. callprogress looked promising, but, alas, as many others have found, it hangs up after timeout seconds. I'll keep digging :). Thanks again everyone, JES James B. MacLean wrote: Hi C F, I am not well versed in this level of telephony or Asterisk, so please bare with me :). My setup is really typical. Bought the digium card with 4 ports. 2 fxs / 2 fxo. The 2 fxo's are connected directly to phones, belong to group 1 according to zapata.conf, and exist as "fxoks=1-2" in /etc/zaptel.conf. The 2 fxs ports are connected to the telco, belong to group 2 according to zapata.conf, and are setup as "fxsks=3-4" in zaptel.conf. Dial(Zap/1/&SIP/[EMAIL PROTECTED],15,r) works as expected, Dial(Zap/2/&SIP/[EMAIL PROTECTED],15,r) works as expected but: Dial(Zap/g2/&SIP/[EMAIL PROTECTED],15,r) Rings once and reports answered to Asterisk. Does this support what you are explaining? I'm honestly confused by how an fxs module operates as an fxo module? Thanks for any more direction you might have, JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP no working in 1/4 installations
Hi folks, New user. Installed latest CVS on 4 PCs. On only one, the fastest one, SIP comes across as if it was getting chopped to pieces. Thought it might be due to the PC being the only AMD processor of the 4, but recompiled with k6 and no difference. IAX seems fine. Tried different SIP clients and from different client PC's. Searched and found a little about sound going silent over time and I am seeing that on an SMP machine I am testing if I connect with SIP, but at least it starts fine when I connect. When I connect to this PC it's as if it is jittering so bad all you get are pops. Since it's a 3200+ and not much else running on it, I expected it to handle a single SIP better ;-). Something obvious I expect but it's a long learning curve (not necessarily steep, but long) and I'm hoping someone can point out what I am missing ;) ? JES smime.p7s Description: S/MIME Cryptographic 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
