[asterisk-users] How to stop the update of astdb?
Hello everybody, I am not using astdb (no func_db and app_db) so I am wondering why asterisk is always updating it. The interval of the update is not constant. Using lsof, I noted the intervals are somewhere between 1 minute to 12 minutes. The output of lsof says that asterisk, atd and crond processes were just active, just after the hard disk changed the state from standby to active/idle. I polled the hard disk's state with hdparm every 1 second using a script. The problem is I have asterisk running on an ancient notebook dedicated only for it. And I want to have its 2.5 hard disk to be standby when asterisk is not handling any calls. I have already moved some of the files (under /var folder) which are regularly updated, into ramdisk. I have another notebook which has the exact same configuration but dedicated for web server. The hard disk is mostly in standby state when it is not handling any web requests. So I want to have it the same for the one for asterisk. I am using debian etch kernel 2.6.23.12 which I compiled particularly for those notebooks. I also tried to have a look on asterisk.c but couldn't find any hints. Do I missed something? Can I prevent asterisk in accessing astdb? Or at least extend the update interval and make the interval more constant? Thanks a lot in advance for your help. Kind regards, Anto ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 - G729 - Have License - No path to translatefrom Zap to IAX2
Forget about this. I rollbacked to 1.2. 1.4 features are quite useless to me without being able to use G729 codec. - Original Message - From: Aryanto Rachmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 28, 2006 9:58 PM Subject: [asterisk-users] 1.4 - G729 - Have License - No path to translatefrom Zap to IAX2 Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756] chan_zap.c: Requested indication 3 on channel Zap/1-1 [Dec 28 21:06:02] WARNING[1734] chan_iax2.c: Received mini frame before first full voice frame . . [Dec 28 21:06:02] WARNING[1736] chan_iax2.c: Received mini frame before first full voice frame [Dec 28 21:06:02] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 answered Zap/1-1 [Dec 28 21:06:02] WARNING[1756] channel.c: No path to translate from Zap/1-1(68) to IAX2/VoIPRakyat-2(256) [Dec 28 21:06:02] WARNING[1756] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/VoIPRakyat-2 I just upgraded to SVN-branch-1.4-r49020M, but doesn't help. I am using TDM400P with one FXO and one FXS. Initially I just compiled and loaded zaptel and wctdm modules. Then I tried to compile and load ztd-eth, ztd-loc, ztdummy, ztdynamic and zttranscode modules as well just to make sure, but that does not help either. I have no issue at all using any other codecs on IAX. There are some threads on this mailing list for similar issue, but mostly pointed out to G729 license. I have one as below: [Dec 28 21:02:52] VERBOSE[1440] logger.c: == G.729 Host-ID: ... [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found license 'G729-' providing 1 channels [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found total of 1 G.729 licenses [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Registered translator 'g729tolin' from format g729 to slin, cost 6 There must be something basic that I missed, maybe the new 1.4 parameters, but I don't know which ones. So please help me out. Thanks a lot in advance. Cheers, Anto ___ --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 ___ --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] 1.4 - G729 - Have License - No path to translate from Zap to IAX2
Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756] chan_zap.c: Requested indication 3 on channel Zap/1-1 [Dec 28 21:06:02] WARNING[1734] chan_iax2.c: Received mini frame before first full voice frame . . [Dec 28 21:06:02] WARNING[1736] chan_iax2.c: Received mini frame before first full voice frame [Dec 28 21:06:02] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 answered Zap/1-1 [Dec 28 21:06:02] WARNING[1756] channel.c: No path to translate from Zap/1-1(68) to IAX2/VoIPRakyat-2(256) [Dec 28 21:06:02] WARNING[1756] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/VoIPRakyat-2 I just upgraded to SVN-branch-1.4-r49020M, but doesn't help. I am using TDM400P with one FXO and one FXS. Initially I just compiled and loaded zaptel and wctdm modules. Then I tried to compile and load ztd-eth, ztd-loc, ztdummy, ztdynamic and zttranscode modules as well just to make sure, but that does not help either. I have no issue at all using any other codecs on IAX. There are some threads on this mailing list for similar issue, but mostly pointed out to G729 license. I have one as below: [Dec 28 21:02:52] VERBOSE[1440] logger.c: == G.729 Host-ID: ... [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found license 'G729-' providing 1 channels [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found total of 1 G.729 licenses [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Registered translator 'g729tolin' from format g729 to slin, cost 6 There must be something basic that I missed, maybe the new 1.4 parameters, but I don't know which ones. So please help me out. Thanks a lot in advance. Cheers, Anto ___ --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] Fedora
I have been using asterisk on FC4, FC5 and now FC6t3 with no irritating problem. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 18, 2006 8:34 PM Subject: [asterisk-users] Fedora Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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] Info about F1000G
Hello Tomislav, I borrowed F1000 from my friend for testing. I am not sure if that is different from F1000G, but I am experiencing the following issues: 1. As a user, it is not easy to get a firmware update as I need to have a service contract. 2. Even with the latest firmware I got from sipgate.de (version 3.80st), I can only have WPA-PSK with TKIP encryption, while I prefer AES. 3. The voice quality is sometimes really bad when using codecs with compression (G729 and G726). No problem with G.711. 4. The battery does not last long, just around 22 hours. I don't have any other issues a part from those. Cheers, Anto - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 8:45 AM Subject: [Asterisk-Users] Info about F1000G Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr ___ --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 ___ --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] Re: Info about F1000G
I forgot one more security issue. I have to set my AP to broadcast the SSID otherwise F1000 phone can not camp on the AP. Security on my wireless network is very important to me. One more info about the wireless encryption. The phone can not camp on my AP when I set my AP to support WPA-PSK with TKIP+AES. It has to be set to support only WPA-PSK with TKIP. Cheers, Anto - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 12:59 PM Subject: [Asterisk-Users] Re: Info about F1000G In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello Tomislav, I borrowed F1000 from my friend for testing. I am not sure if that is different from F1000G, but I am experiencing the following issues: 1. As a user, it is not easy to get a firmware update as I need to have a service contract. 2. Even with the latest firmware I got from sipgate.de (version 3.80st), I can only have WPA-PSK with TKIP encryption, while I prefer AES. 3. The voice quality is sometimes really bad when using codecs with compression (G729 and G726). No problem with G.711. 4. The battery does not last long, just around 22 hours. I don't have any other issues a part from those. Cheers, Anto Hi everybody! Thank you all for information's that you have provide to me. Now I have pretty clear picture what to expect from this phone. Have a nice day! -- Tomislav Parcina tparcina#lama.hr ___ --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 ___ --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] dtmfmode=auto, but doesn't work
Hello everybody,I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added in CVS HEAD on Sep 6 2005 and is not available in Asterisk 1.0.x. Whydoes it not work as the wiki said? I am running asterisk SVN-branch-1.2-r9609M under FC5-t2 (2.6.15-1.1928_FC5smp). Cheers, Anto ___ --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] Zap, Caller ID problem
Do you have the following set in your zapata.conf? callerid=asreceived - Original Message - From: KaveH Aasaraai [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 12, 2006 10:25 AM Subject: [Asterisk-Users] Zap, Caller ID problem Dear All, I've got a weird problem with my asterisk box which has fxo interfaces (TDM400). Well, the problem is that the interface answers the call, but no caller id is being received. Also, sometimes this error happens: fsk_serie made mylen 0 Any idea what is going on? Thanks, Kaveh __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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] dtmfmode=auto, but doesn't work
Thanks a lot Kevin, I am aware that inband DTMF does not work over G.729 codec. So in this case my provider does not offer RFC2833. I can not do anything about this, can I? Or is there anyway to simulate inband DTMF over any other codecs, but G.711a or G711u? Cheers, Anto - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 12, 2006 5:31 PM Subject: Re: [Asterisk-Users] dtmfmode=auto, but doesn't work Aryanto Rachmad wrote: Why does it not work as the wiki said? It does work exactly as the wiki said. The SIP peer did not offer to send/receive RFC2833, so we assume it wants to use inband DTMF. However, inband DTMF _does not work_ over G.729 codec, so you get a warning. ___ --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 ___ --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] How to remove first ring tone on FXO?
Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --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] How to remove first ring tone on FXO?
Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, January 29, 2006 9:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] How to remove first ring tone on FXO?
Thanks a lot Dean, I think thereis a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you pleaseletme know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, 29 January 2006 10:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, Ijusttried that, but itdoesn't help. There is still one ring tone produced before asterisk executes Answer(). And thereis nocaller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto RachmadSent: Sunday, January 29, 2006 9:40 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port,asterisk producesone ring tone just beforeit executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] How to remove first ring tone on FXO?
Hello CF, I thought that asterisk generated that first ring tone. I didn't think further, especially about what the caller's switching centre is doing when it gets an instruction to reach my number. You are obviously right. That switch will notify the caller (alerting) as soon as it gets a connection confirmed message from my switching centre. And I definitely can not avoid that. Thanks a lot for explaining this. Cheers, Anto - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 29, 2006 5:32 PM Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? You are wrong, there is no way you can remove the ring, since the ring is something that the callers equipment is generating to the caller, and NOT asterisk. The most you will able to accomplish will be to have just one ring before asterisk picks up. By setting usecallerid to no all you are doing is telling asterisk don't wait for callerid, but since you are using POTS, 2 things will always happen that you can't control: 1. At least part of a ring has to be delivered to Asterisks' FXO port, so that Asterisk knows that there is an incoming call, because inband signalling is used, there is no other way for asterisk to know there is an incoming call. 2. The calling party will always hear at least one ring even if asterisk happens to pick up the line - by mistake - before any ring voltage, because the switch that the line is connected to has already sent a ring indicator to the remote switch, and the remote switch has already started playing the ring tone to the caller, similar to what the playtones(ring) does in Asterisk. Even if it happens to be that you were able to get it once to NOT ring to the caller, it was by mistake that the timing worked out that way, it has nothing to do with what you set in Asterisk. If you have a PRI then you can do that, since if you use Answer as the first command, ring indicator is never sent down the line. On 1/29/06, Aryanto Rachmad [EMAIL PROTECTED] wrote: Thanks a lot Dean, I think there is a way to remove that ring tone and also still have the caller ID from the incoming call. I have been trying to find that on zaptel.c, chan_zap.c and pbx.c, but I could not find that. Could you please let me know which part of the codes handling that? Cheers, Anto - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 4:23 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? Anto, Callerid delays answer until after the first ring, I would suggest you are either not subscribing to your telco for caller id or similar. The advice you got was correct. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, 29 January 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to remove first ring tone on FXO? Thanks Alexander, I just tried that, but it doesn't help. There is still one ring tone produced before asterisk executes Answer(). And there is no caller ID being forwarded to the destination channel, which actually I need. That is why I have usecallerid set to yes. Cheers, Anto - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 29, 2006 3:54 PM Subject: RE: [Asterisk-Users] How to remove first ring tone on FXO? It is waiting for the CalledID information. Set usecallerid=no and that should do it for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad Sent: Sunday, January 29, 2006 9:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to remove first ring tone on FXO? Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented #define RINGBEGIN on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
Re: [Asterisk-Users] RE: No audio? Update your Asterisk
Hello Brent, I think I have experienced the same issue. I had problem with annoying echo on my FXS port (TDM400P). I was using SVN-branch-1.2-r8445. I implemented James Harper's preload patch by hand yesterday night. I was happy with the result. I didn't have any audio issue until my wife complained this morning as she couldn't hear anything. I switched off my asterisk and went to the office. I just found out a few hours ago that there is a critical issue, so I updated my asterisk to SVN-branch-1.2-r8666. I got the audio back, but I also got the annoying echo again. So I went back to Asterisk 1.2.1. The audio is alright and no echo on my FXS. Cheers, Anto - Original Message - From: Brent Torrenga [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 5:57 PM Subject: [Asterisk-Users] RE: No audio? Update your Asterisk Yeah, got two whacked 1.2.2 servers here. Updated to 1.2.3 from ftp source, and a-ok. HOWEVER, echo cancellation seems to be non-existant in our TDM card?!? I recompiled/installed zaptel 1.2.2, no effect. Even tried different cancellers. Anyone else experience this after upgrading to 1.2.3? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --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 ___ --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] Reducing echo on FXS port (SOLVED - Big THANKS to James Harper)
Hello All, The solution which is good enough for me at the moment, is James Harper's echo preload patch. The echo is reduced to the minimum from the start of the call. I just need to reduced the rxgain to make it more unnoticeable. I implemented on SVN-branch-1.2-r8445 by hand yesterday, but I got no audio this morning. After updating to SVN-branch-1.2-r8666, the audio came back, but also the echo. So I decided to use Asterisk 1.2.1 and put in the patch. Cheers, Anto - Original Message - From: Aryanto Rachmad To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 23, 2006 12:13 AM Subject: Re: [Asterisk-Users] Reducing echo on FXS port Hello Giovanni and everybody, Thanks a lot for your suggestion. Unfortunately, that does not help. With READ_SIZE=16, I got vibrating voice on the speaker phone. With READ_SIZE=80, the voice came back to normal and the echo is more reduced but still noticeable. I finally changed it back to 160 as it seems to affect a lot of things in chan_zap. I am sorry that I didn't properly explain the setup I have. I think this issue happens because I use wireless handset. Here is my actual setup: --- out | | --- | |Wireless handset -- Base phone -- FXS (TDM400P) -- | ZAP Channel | | Asterisk | (Mic muted) | | --- | | --- in ^ Monitor() The delay between the original voice going out from asterisk to the phone, and its echo coming back to asterisk is about 104 ms, assuming that Monitor() application wrote both files at exactly the same time. How did I find that? I loaded the "in" and "out" files created by Monitor() application into Audacity (audacity.sourceforge.net). I could not find any other method to find this delay. Does anyone know a better and more accurate method? When I did the same thing using X-Lite with below setup: --- out | | --- | |PC (X-Lite) -- | SIP Channel | | Asterisk |(Mic muted) | | --- | | --- in ^ Monitor() There is no sound at all on the "in" file created by Monitor() application, indicating that there is no echo at all as the microphone on X-Lite was muted. Since the echo on FXS is consistent, there must be a way to eliminate it. The question is how? I think it can not be done only by changing the configuration parameters related to echo, especially with this huge delay. Do you have any other suggestions? Cheers, Anto ___ --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] Asterisk 1.2.3 Released - Critical Update
Totally agree! :) - Original Message - From: Matt Riddell (IT) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 26, 2006 1:30 AM Subject: Re: [Asterisk-Users] Asterisk 1.2.3 Released - Critical Update Wiley Siler wrote: I am going to assume the best and hope it was a an issue of testing code missed at release. Or you could just read the code...it is Open Source... :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ --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] Voipbuster/voipstunt -- what a crap service
Didn't you read this from their QA? I want to configure my own IAX/SIP device for calling with VoipBuster, is that possible? It is possible to use your own IAX/SIP device, however we do not support it. We advise you to use SIP-Discount instead. Do you have the same problem when you use their softphone? If not, why complaining. The call to the UK is free only for VoIPstunt - Original Message - From: RumaTech [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 26, 2006 6:19 AM Subject: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got authentication request (401) on unknown BYE to 'sip:[EMAIL PROTECTED];tag=c9ebef50c90078c2c93eddc243d7352d6e04' Secondly, they charged me for calls to UK that was supposed to be free. And their customer service does not respond at all. Do they have a phone number I can call? Thanks, Rudolf ___ --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 ___ --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] Reducing echo on FXS port
Hello Giovanni and everybody, Thanks a lot for your suggestion. Unfortunately, that does not help. With READ_SIZE=16, I got vibrating voice on the speaker phone. With READ_SIZE=80, the voice came back to normal and the echo is more reduced but still noticeable. I finally changed it back to 160 as it seems to affect a lot of things in chan_zap. I am sorry that I didn't properly explain the setup I have. I think this issue happens because I use wireless handset. Here is my actual setup: --- out | | --- | |Wireless handset -- Base phone -- FXS (TDM400P) -- | ZAP Channel | | Asterisk | (Mic muted) | | --- | | --- in ^ Monitor() The delay between the original voice going out from asterisk to the phone, and its echo coming back to asterisk is about 104 ms, assuming that Monitor() application wrote both files at exactly the same time. How did I find that? I loaded the "in" and "out" files created by Monitor() application into Audacity (audacity.sourceforge.net). I could not find any other method to find this delay. Does anyone know a better and more accurate method? When I did the same thing using X-Lite with below setup: --- out | | --- | |PC (X-Lite) -- | SIP Channel | | Asterisk |(Mic muted) | | --- | | --- in ^ Monitor() There is no sound at all on the "in" file created by Monitor() application, indicating that there is no echo at all as the microphone on X-Lite was muted. Since the echo on FXS is consistent, there must be a way to eliminate it. The question is how? I think it can not be done only by changing the configuration parameters related to echo, especially with this huge delay. Do you have any other suggestions? Cheers, Anto - Original Message - From: Giovanni Miano To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, January 15, 2006 2:27 PM Subject: Re: [Asterisk-Users] Reducing echo on FXS port TryIn chan_zap.c change the following line:#define READ_SIZE 160to#define READ_SIZE 16In zapata.confjitterbuffers=40This will also increase system load by a factor of 10. ___ --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] jitterbuffer on zap channel
Hello All, What is the advantage of jitterbuffer on zap channel? do you have any suggestion on the setting for home usage? Is there anydisadvantage in using it? Cheers, Anto ___ --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] Reducing echo on FXS port
Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 -- FXS (TDM400P) -- Asterisk -- SIP GW -- PSTN -- Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echoto maximum with the following settings for my FXS port on zapata.conf: rxgain=-8.0txgain=2.0echocancel=256echotraining=500 Butit isstill not entirely eliminated as westill sometimes hear the last syllables, with the level ofmaybe 5% of the original sound. What I did was just playing around with the values of those parameters, use ztmonitor to have the FXS rx/tx signalvisualised and use only my ears to check it. I think my ears are fine :), as I dothis because my friends complain about the echo they hear. Does anybody know a better method tofind the best value forthose parameters? There is no echo on phone2 when Iusesoftphone like this: PC(X-Lite) -- Asterisk -- SIP GW -- PSTN -- Phone2 The following is the version of asterisk I am using: CLI show version Asterisk SVN-branch-1.2-r7999 built by root @ atvie-asterisk on a i686 running Linux on 2006-01-13 06:15:02 UTC And I set the echo canceller in zconfig.h to ECHO_CAN_MG2. Cheers, Anto ___ --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] IAX problem - Bug or Compatibility issue?
Hello All, I am looking for more thorough debug than the one provided by the command "iax2 debug". Could anybody point me a good documentation about this? I have a issue with IAX connection. Sometimes it stucked.If so, I have to restart my asterisk through CLI command"restart now". Comparing the debug messages of working and non working sequences, I have noticed that when it does not work, the following debug messages are missing: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 01581ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01581ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] -- IAX2/sipdiscount_outbound-16385 is making progress passing it to Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4791 zt_indicate: Requested indication 14 on channel Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4857 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/1-1Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 01732ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Dec 30 17:12:31 DEBUG[12569]: chan_iax2.c:6653 socket_read: Ooh, voice format changed to 4Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01732ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] I have a few questions, especially about the following message: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) 1. Is the number 14 in (14?), in decimal or hexadecimal? 2. If that is in decimal, why isit not translated into its descriptions, i.e. Call Progress, according to the IAX2 protocol document I have (Internet-Draft, Expires: July 5, 2005). 3. Why isthat numberquestion marked? Is it because asterisk was not sure? 4. If asterisk was not sure, so sometimes it decodes the message sometimes it could not, is there any debug to confirm this? Or, am I looking at the wrong place? Which maybe the problem is so obvious and I missed that? I am running asterisk on IBM xSeries 330 with the following detail: CLI show versionAsterisk 1.2.1 built by root @ atvie-asterisk on a i686 running Linux on 2005-12-28 07:52:36 UTC# uname -aLinux atvie-asterisk 2.6.14-1.1653_FC4smp #1 SMP Tue Dec 13 21:46:01 EST 2005 i686 i686 i386 GNU/Linux Please find also below the detail of IAX debug messages. Cheers, Anto MESSAGES WHEN IAX DOES NOT WORK -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulawTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00057ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00080ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3699 find_tpeer: Created trunk peer for '213.61.187.147:4569'Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3725 iax2_trunk_queue: Expanded trunk '213.61.187.147:4569' to 6400 bytesRx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] --- Some messages are missing hereTx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10016ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10008ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 10262ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] CAUSE CODE : 0 MESSAGES AFTER ISSUING "CLI restart now" command -- Call accepted by 213.61.187.157 (format ulaw) -- Format for call is ulawTx-Frame Retry[-01] --
Re: [Asterisk-Users] IAX problem - Bug or Compatibility issue?
Nevermind. I think I found it. I didn't realise that Ethereal can decode IAX2 protocol. When IAX does not work,sometimes I got the following message as well: DEBUG[2213] chan_iax2.c: Immediately destroying 16384, having received INVAL What else can we say about that a part frommy asterisk receivedframe with INVAL(Invalid call) or in other word incompatible frame? I also found a very quick and dirty workaround which is enough for me at the moment. - Original Message - From: Aryanto Rachmad To: asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 6:38 PM Subject: [Asterisk-Users] IAX problem - Bug or Compatibility issue? Hello All, I am looking for more thorough debug than the one provided by the command "iax2 debug". Could anybody point me a good documentation about this? I have a issue with IAX connection. Sometimes it stucked.If so, I have to restart my asterisk through CLI command"restart now". Comparing the debug messages of working and non working sequences, I have noticed that when it does not work, the following debug messages are missing: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) Timestamp: 01581ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01581ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] -- IAX2/sipdiscount_outbound-16385 is making progress passing it to Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4791 zt_indicate: Requested indication 14 on channel Zap/1-1Dec 30 17:12:31 DEBUG[12600]: chan_zap.c:4857 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/1-1Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 01732ms SCall: 00052 DCall: 16385 [213.61.187.157:4569]Dec 30 17:12:31 DEBUG[12569]: chan_iax2.c:6653 socket_read: Ooh, voice format changed to 4Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01732ms SCall: 16385 DCall: 00052 [213.61.187.157:4569] I have a few questions, especially about the following message: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: (14?) 1. Is the number 14 in (14?), in decimal or hexadecimal? 2. If that is in decimal, why isit not translated into its descriptions, i.e. Call Progress, according to the IAX2 protocol document I have (Internet-Draft, Expires: July 5, 2005). 3. Why isthat numberquestion marked? Is it because asterisk was not sure? 4. If asterisk was not sure, so sometimes it decodes the message sometimes it could not, is there any debug to confirm this? Or, am I looking at the wrong place? Which maybe the problem is so obvious and I missed that? I am running asterisk on IBM xSeries 330 with the following detail: CLI show versionAsterisk 1.2.1 built by root @ atvie-asterisk on a i686 running Linux on 2005-12-28 07:52:36 UTC# uname -aLinux atvie-asterisk 2.6.14-1.1653_FC4smp #1 SMP Tue Dec 13 21:46:01 EST 2005 i686 i686 i386 GNU/Linux Please find also below the detail of IAX debug messages. Cheers, Anto MESSAGES WHEN IAX DOES NOT WORK -- Call accepted by 213.61.187.147 (format ulaw) -- Format for call is ulawTx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00057ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 00080ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3699 find_tpeer: Created trunk peer for '213.61.187.147:4569'Dec 30 17:04:25 DEBUG[12488]: chan_iax2.c:3725 iax2_trunk_queue: Expanded trunk '213.61.187.147:4569' to 6400 bytesRx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00080ms SCall: 00070 DCall: 16384 [213.61.187.147:4569] --- Some messages are missing hereTx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10008ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10016ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10016ms SCall: 16384 DCall: 00070 [213.61.187.147:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10008ms SCall: 00070 DCall: 16384 [213.61.187.147:4569]Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK T
Re: [Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router
Hello Robert, I have this following setting on my WRT54GS: # RTP ports iptables -t nat -A PREROUTING -i $WAN -m udp -p udp --dport 1:2 -j DNAT --to-destination $ASTERISK_IP iptables -A FORWARD -i $WAN -o $DMZ -m udp -p udp --dport 1:2 -d $ASTERISK_IP -j ACCEPT # IAX port iptables -t nat -A PREROUTING -i $WAN -m udp -p udp --dport 4569 -j DNAT --to-destination $ASTERISK_IP iptables -A FORWARD -i $WAN -o $DMZ -m udp -p udp --dport 4569 -d $ASTERISK_IP -j ACCEPT # SIP port iptables -t nat -A PREROUTING -i $WAN -m udp -p udp --dport 5060 -j DNAT --to-destination $ASTERISK_IP iptables -A FORWARD -i $WAN -o $DMZ -m udp -p udp --dport 5060 -d $ASTERISK_IP -j ACCEPT Cheers, Anto - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 7:37 AM Subject: [Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router Can someone please send me your iptables rules for forwarding SIP/RTP udp to your * server? I tried this but I think I need more rules like DNAT or something... iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 5060 -d $ASTERISK_IP --dport 5060 -j ACCEPT iptables -A FORWARD -i $EXT_IF -o $INT_IF -p udp -m udp --sport 1:2 -d $ASTERISK_IP --dport 1:2 -j ACCEPT ___ --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 ___ --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] problem with tdm400 fxo
Hello Filippo, What revision of TDM400P do you have? Is it REV I? I have REV I and had the same problem before. I had problem when I connected the FXO port to the all plug using 4 wires phone cable. It turned out that the RJ11 port of FXO or FXS in REV I, does not work when pin 1 and pin 4 are connected to something. My problem was solvedafter I changed the cable to 2 wires phone cable. Cheers, Anto - Original Message - From: Filippo Carone To: asterisk-users@lists.digium.com Sent: Friday, December 23, 2005 11:22 PM Subject: [Asterisk-Users] problem with tdm400 fxo Hi,I'm experiencing a very weird behaviour with my tdm400 with two fxo and one fxs modules. I setup my current configuration at home, I tried it and it works flawlessly. I moved the computer to my office and plugged the fxo to the wall plug, but when I tried to call I got a busy signal. I attached the same wire to a phone, I called again and the phone rang. I tried with both the fxo ports, but I always got a busy signal and on the CLI Asterisk doesn't notice the incoming call at all. Outgoing calls do not work either. So I moved again the computer to the home of a friend, and it there it works too as it does at my place. When I plugged the TDM at the office no other phone was plugged in the whole structure.I'm really puzzled and I don't know why it is behaving this way. Any hints? Cheers,fc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Asterisk on Solaris SPARC
Hello everybody, Is there anybody successfully have Asterisk running particularly on Sun Fire V100 (64 bits) with Solaris 9? Any hints and suggestions would be much appreciated. Cheers, Anto ___ --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] TDM400P recognised as Network controller: Unknown device
Hello everybody,I have been googling for hours and also searchedon http://www.voip-info.org/wiki-Asterisk, but I still can not find anyinformation for the problem I have. SoI hope one of you could help me out.I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels though, no zaptel installed. As I wanted to explore more, I bought a TDM400P development kit (TDM11B) from an authorised Asterisk reseller in Germany. After I updated my Asterisk (make update) and installed zaptel last week (27 Sep 2005), here is what I got:# lspci -v01:05.0 Network controller: Unknown device e159:0001 Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 100, IRQ 209 I/O ports at 2400 [size=256] Memory at efffe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2# dmesgModule 0: Installed -- AUTO FXS/DPOModule 1: Not installedModule 2: Not installedModule 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Both LEDs of the FXO and FXS modules are illuminated. I guess that is a normal state.Is the status of "Unknown device" a normal status?From what you have experienced, is there any issue with revision I? I know that there are problems on the cards with revision E or F, so I don't want to waste my time trying to configure the card, which maybe in the end I have to return the card to be replaced as well.Do you think this is just an issue of the driver (zaptel) or something else? Do you have any hints on what should be changed or modified?I sent an email to Digium support and got only a reply like this:"Although the card is being shown as an 'Unknown Device', it should still work properly."To be honest, I am not happy with that answer.FYI, I have installed Asterisk (CVS HEAD - 27 Sep 2005) on IBM xSeries 330 (8654-51Y) running Fedora Core 4 (kernel 2.6.12-1). Thanks in advance for youranswers.Kind regards,Anto ___ --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
