Re: [Asterisk-Users] PRI in Shanghai China
陈帆 wrote: steven,, u not understand the situation,,, there only have 30 channel for voice,, the other channel resiverd for control channel.. good luck for you. You seem to have a problem counting. 1 to 15, plus 17 to 31 == 30 voice channels. Steve On 5/10/06, *Steve Underwood* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: 陈帆 wrote: there have only 30 channel for E1,, He has defined the correct 30 channels for his E1. On 5/10/06, *Steve Underwood* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Edwin Lam wrote: hi folks. does any one have experience setting up E1 PRI in Shanghai, China? it works fine when we use SIP phone to dial out, however when using forward function on the same phone, it seems like it's dialing out but there's actually no respond from the phone company (China Telecom) and eventually the dial command will timed out. here's our PRI portion of zapata.conf: switchtype=euroisdn nsf=none pridialplan=unknown internationalprefix = 00 nationalprefix = 0 localprefix = privateprefix = unknownprefix = signalling=pri_cpe hidecallerid=no usecallingpres=yes callerid=asreceived channel = 1-15 channel = 17-31 any clues? China Telecom can be very fussy about getting the TON and NPI values right, and some installations simply fail to respond to things they don't like. As with all telcos, if things don't work properly asking China Telecom to help is usually a frustrating task. Steve -- Jeffery ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery iaxtel Num: 1-700-576-1311 fwdnet Num: 728150 ___ --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] Which is the best fax-modem for testing ?
Lee Howard wrote: Olivier Krief wrote: For example, it seems that Brother 8360P uses Super G3 mode. Is there a fax-modem offering such capability so that I could easily check if I still cannot hangup when I enable or disable Super G3 mode ? MultiTech 5634-series and MainPine RockForce fax modems (Agere chipset) support SuperG3. You'd run these with HylaFAX, for example, and not Asterisk. It is worth pointing out that the V.34 modems have almost no chance of achieving V.34 speeds if you go: PSTN-analogue line-asterisk-FXS port-modem if you go PSTN-digital line-asterisk-FXS port-modem performance will depend on the FXS port, and any internal timing issues. With a TDM400 card its fairly unlikely to work. With a channel bank connected to a port on the same digital card that connects to the PSTN chances are high. The problem with the PSTN-analogue line-asterisk-FXS port-modem path is signal degradation through the extra analogue-digital-analogue step is too much for V.34. For FAX modems up to V.29 it is no problem. For V.17 is tends to work if the port quality is good. Steve ___ --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] no ringtone
Hi, I have a queue that plays music when a call comes in. To be able to do that I need to Answer() the call first. After a timeout in this scenario the call should be transfered to an extension using a GoTo statement to the extensions context. The problem is that as soon as asterisk Answers the call it can not play a ringtone (or other tones) back to the original caller when executing a Dial . Is it possible to transfer the call in the dialplan or how can I solve this? It doesn't help to use the r option in Dial(SIP/104|30|r). This is the output from asterisk where indication 3 according to the source code is ringing: May 21 11:03:10 WARNING[12188]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/XX-c52d' /urban ___ --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] Events offered by
Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix ___ --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] Any IP phones with pro-audio connections?
I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is picked up. The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should have to (something like +8 level on the output channel from the soundboard into the AutoHybrid), but that could be the ATA. Hope this helps,GarthOn 5/21/06, Colin Anderson [EMAIL PROTECTED] wrote:Snom 200 if you can get ahold of them has standard 1/8 headset jacks you would use the Monitor() app to record.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ]Sent: Saturday, May 20, 2006 5:52 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Any IP phones with pro-audio connections?The new grandstream video phone has rca-style audio jacks Paul HalesTechnical ManagerAsteriskITwww.asteriskit.com.au- Original Message -From: Julien Goodwin [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSent: Saturday, May 20, 2006 9:30 PM Subject: [Asterisk-Users] Any IP phones with pro-audio connections? Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8 jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien ___ --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 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 unlock old SCCP Cisco 7960 ?
Hi,Thanks to Cory's and HTH's help (spell ? black magic ?), I could at last unlock Network Configuration's menu.To unlock, I pressed Settings key then a combination of # and * keys from dialpad and it worked. I really thought I've tried every possible combination before writing to this list but this time it worked.My phone could reach Registration Rejected stage and is not looping on Configuring IP anymore. Thanks for all ___ --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] no ringtone
Urban wrote: May 21 11:03:10 WARNING[12188]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/XX-c52d' You don't have a /etc/asterisk/indications.conf -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Upgrade 7960 from SCCP 3.0 to SIP 7.5
Hi,I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ?From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2 , I got the following: Copy the desired binary image from Cisco.com to the root directory of the TFTP server. Specify the image in the configuration file image parameter for the protocol to which you are converting (load_information for SCCP or image_version for SIP). Remove any protocol configuration files that are not used for the specified protocol. Firmware versions are P003F300 (Application Load ID) and PC030300 (Boot Load ID).In TFTP root directory, the following files are present:OS79XX.TXTP003-07-5-00.bin P003-07-5-00.sbnP0S3-07-5-00.bin P0S3-07-5-00.sbn P0S3-07-5-00.loadsSIPMAC Addres.cnfSIPDefault.cnfOn boot, I can see in my TFTP logs the phone is served an OS79XX.TXT file which now holds P0S3-07-5-00 content.From TFTP logs, I can see my phone is then asking for P0S3-07-.bin file which doesn't exist in my TFTP directory.Next it asked for SEPMAC Addres.cnf and SEPDefault.cnf.Both files don't exist but SIPMAC Addres.cnf and SIPDefault.cnf do exist.In SIP Default.cnf image_version=P0S3-07-5-00 is included..When I change P0S3-07-5-00 to P003-07-5-00 in OS79XX.TXT file, it directly asked SEPMAC Addres.cnf and SEPDefault.cnf failing to ask for any .bin file. My first question is :What should be written is OS79XX.TXT if I want to upgrade to SIP ?P0S3-07-5-00 ?P0S307500 (with a symbolic link to P0S3-07-5-00 in TFTP root directory) ? P003-07-5-00 ?Regards ___ --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] Configuring a TDM400P with one FXS port
Hi, I have a TDM400P and it works, Send us the output of your dmesg command. Regards. --- M.Hockings [EMAIL PROTECTED] a écrit : In my attempt to setup a single FXS line I have been following the instructions for Telephony Card Drivers on the asteriskdocs.org site. I have managed to checkout, make and install the zaptel code and can load the zaptel module but when I attempt to load the wcfxs module it tells me: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm On the card the single FXS module is in the position at the back of the TDM400P (i.e., closest to the connectors) In the /etc/zaptel.conf file I have put the following at the bottom of the file: fxoks=1 loadzone=us defaultzone=us I have also tried fxoks=4 with the same results other than that the channel number changes in the message. Any idea what I am configuring incorrectly ? Mike ___ --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 ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] transfer outside of a call?
Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Thank you, Juraj. ___ --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 monitor DTMF tones in a call?
Quoting Moises Silva [EMAIL PROTECTED]: I downloaded and compiled this trunk version - Asterisk SVN-trunk-r28970. The DTMF events show up in the logging system after I configured logger.conf to output them, but they are not showing up in the Events. On checking the SVN for the 6082 patch I saw a branch ../team/jcollie/bug6082. I don't know what revision that branch is based on. Will compiling that branch give me the facility? I am not that familiar with the SVN workings, but if you give the instructions to follow and may be a revision number or some other parameters to work with I will be able to do the rest myself. You can check that info in www.asterisk.org or voip-info.org If you have problems applying the patch let me know, may be I can make you a patch for the 1.2.7.1 specially. Regards On 5/19/06, Obelix [EMAIL PROTECTED] wrote: Quoting Moises Silva [EMAIL PROTECTED]: Hi, I am ready to try out this patch, both PlayDTMF and SendDTMF and want to know which branch I should work from. I am not quite experienced with compiling from SVN directly and would like to know whether to download the latest 1.2.7.1 and apply the patch to it or use the latest from SVN. Can you give me a list of commands I should apply to SVN? /Obelix I have uploaded a patch for some manager events that allow to know when DTMF has been received or sent. Please take a look at this: http://bugs.digium.com/view.php?id=6082 and if you can, test it and report feedback. Im having problems to call the attention of bug marshalls for comitting this change. I think this week i will enter to IRC in asterisk-dev to try to make that bugmarshalls pay attention to it. Best Regards On 4/30/06, Obelix [EMAIL PROTECTED] wrote: Is there a way to monitor the DTMF tones on a channel? I have a prepaid application working in asterisk. When the user dials a call and wants to cancel the call before it is answered, there is now way to do it without hanging up and redialling the access number. Is there way to monitor a sequence of DTMF tones and cancel the call? If I use a SIP gateway or proxy rather than dial asterisk directly will that be possible? ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Skill-based routing
Hello, does anybody know about an existing skill-based routing solution for asterisk? I found only some theoretical documents on voip-info.org. I would like to have finer control over who can get which call in which order. Example: Several operators with several topics. Each operator may have a given knowledge-base for given topic. Topics may be weighted in question of complexity as well. Some operators may be experts, able to answer hard questions, while some may only make some basic tasks [e.g. activate/deactivate services, etc.]. Of course we don't want to give a simple question to an expert if there are other people to be able to answer it [if another question comes in which is complex, let's have the expert free if possible]. Classification of questions may be done with IVR. The ACD should have also waiting queues with MOH. As you can see, there is no real concept yet, what to do, but from this I can see, that the given complicated ACD cannot be done in app_queue (or am I wrong?). I was thinking about rewriting app_queue to let communicate with an external application about routing decision. Thus an external script [e.g. daemon process listening on UDP] can easily decide where to route the call. In a high-level language it won't be a problem. Of course, that external application must know the current state of all agents/operators, weights, etc. Another approach is to use a combination of - Manager API: to collect states and give some commands like Redirect - FastAGI: to make the routing decision - dialplan: to play IVR, MOH, call the FastAGI, analyze results, route in loop Of course there are several open question with this solution. I'm searching for an optimal solution which can be very simple and with some programming [expanding the external operative application] can be very complex, allowing different skill-based routing systems. Any help, idea appreciated. Kind regards, Tamas ___ --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] WiFi VoIP Handsets..
2006/5/17, Cory Andrews [EMAIL PROTECTED]: I think around Q3/Q4 of this year, you'll see some very interesting newproducts which incorporate DECT for wireless.For consumer products withlimited mobility, it seems to make a bit more sense than WIFI. Cory,Which DECT products are you specifically thinking of ?Cheers ___ --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] WiFi VoIP Handsets..
2006/5/16, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi,I have got hold of a Nokia E60 myself but could yet connect to any SIPserver. WLAN works fine and i am able to browse thru my SMC wirelessLAN. If some one has successfuly done the SIP configs, i would like to get the detailsThanks in advance.DanDan,I've just read E60 is available since last monday (475 Euro, tax excluded !).But it seems, as you mentioned, it doesn't connect to SIP proxy though I saw presentations from Alcatel and Cisco, specifically mentionning this product could connect to their Call Managers. Did you get any support from Nokia explaining its compliance to Asterisk ?Regards ___ --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] Skill-based routing
Simple solution. Use different queues for skillsets, add agents with penalties based on their skillset. 1 means expert, 100 means you just want the call answered if nobody with a lower penalty is available. Use an IVR to direct the caller to the correct queue (topic). Penalties can be set through the manager interface with a small patch. I did not write the patch and therefore cannot supply it but it is possible and took a good developer about ten minutes to write. Thanks, Steve Tamas wrote: Hello, does anybody know about an existing skill-based routing solution for asterisk? I found only some theoretical documents on voip-info.org. I would like to have finer control over who can get which call in which order. Example: Several operators with several topics. Each operator may have a given knowledge-base for given topic. Topics may be weighted in question of complexity as well. Some operators may be experts, able to answer hard questions, while some may only make some basic tasks [e.g. activate/deactivate services, etc.]. Of course we don't want to give a simple question to an expert if there are other people to be able to answer it [if another question comes in which is complex, let's have the expert free if possible]. Classification of questions may be done with IVR. The ACD should have also waiting queues with MOH. As you can see, there is no real concept yet, what to do, but from this I can see, that the given complicated ACD cannot be done in app_queue (or am I wrong?). I was thinking about rewriting app_queue to let communicate with an external application about routing decision. Thus an external script [e.g. daemon process listening on UDP] can easily decide where to route the call. In a high-level language it won't be a problem. Of course, that external application must know the current state of all agents/operators, weights, etc. Another approach is to use a combination of - Manager API: to collect states and give some commands like Redirect - FastAGI: to make the routing decision - dialplan: to play IVR, MOH, call the FastAGI, analyze results, route in loop Of course there are several open question with this solution. I'm searching for an optimal solution which can be very simple and with some programming [expanding the external operative application] can be very complex, allowing different skill-based routing systems. Any help, idea appreciated. Kind regards, Tamas ___ --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] Any IP phones with pro-audio connections?
Wow. That certainly looks like a definitive solution. I'd have thought that any PC & soft phone with cheap USB audio I/F would provide at least simplex audio. I've seen USB I/Fs with XLR +4 dbm output at music shops for $40. Michael --Original Message Text--- From: Garth Summey Date: Sun, 21 May 2006 11:48:42 +0200 I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is "picked up". The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should have to (something like +8 level on the output channel from the soundboard into the AutoHybrid), but that could be the ATA. Hope this helps, Garth On 5/21/06, Colin Anderson [EMAIL PROTECTED] wrote:Snom 200 if you can get ahold of them has standard 1/8 headset jacks you would use the Monitor() app to record. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] Sent: Saturday, May 20, 2006 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any IP phones with pro-audio connections? The new grandstream video phone has rca-style audio jacks Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au - Original Message - From: "Julien Goodwin" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Saturday, May 20, 2006 9:30 PM Subject: [Asterisk-Users] Any IP phones with pro-audio connections? Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8" jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien ___ --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 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] update or add DID's to directory Assistance
I need some help getting our DIDs updated in the directory assistance and also the caller ID cname that is displayed. Does anyone know where to go to do this so the major carriers will get this info? I have found www.listyourself.net but I am hesitating to submit info to someone I have never heard of and there is really no contact info on the there site besides an email address. Any info would be great? Thanks Rick ___ --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] WiFi VoIP Handsets..
HmmmI have a 480i-CT. Does this mean that I might be able to add third party DECT handsets? Or just the matching Aastra handsets? Michael --Original Message Text--- From: Dovid Bender Date: Sat, 20 May 2006 12:54:54 -0700 (PDT) Cory, Do you have the Nokia E70 and or the E60 ? If not are you guys gona get it in anytime soon ? Dovid Cory Andrews [EMAIL PROTECTED] wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --0-523538568-1148154894=:96965-- ___ --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] WiFi VoIP Handsets..
I know it supports additional Aastra handsets, up to 10 I believe. As for 3rd party, not sure about that... Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Michael Graves To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, May 21, 2006 11:19 AM Subject: RE: [Asterisk-Users] WiFi VoIP Handsets.. HmmmI have a 480i-CT. Does this mean that I might be able to add third party DECT handsets? Or just the matching Aastra handsets?Michael--Original Message Text---From: Dovid BenderDate: Sat, 20 May 2006 12:54:54 -0700 (PDT)Cory,Do you have the Nokia E70 and or the E60 ? If not are you guys gona get it in anytime soon ?DovidCory Andrews [EMAIL PROTECTED] wrote:The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote,wireless handsets via DECT.Cory AndrewsExecutive Vice President++VoIPSupply.comPBXSelect.com++454 Sonwil DriveBuffalo, NY 14225voice - 800.398.VoIP X3402fax - 716.630.1548e - [EMAIL PROTECTED]m - 716.907.4059aim - B2Cory-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of WipeOutSent: Tuesday, May 16, 2006 10:38 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] WiFi VoIP Handsets..James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. JamesThanks for all the replies..James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for..Any pointers to recommended DECT VoIP phones?___--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__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --0-523538568-1148154894=:96965-- ___--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] British English voice files are ready for download
On 19 May 2006, at 17:05, Mark Phillips wrote: Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} It is great that there is an extra (British) choice for asterisk sounds. I downloaded the SLIN and I have a couple of remarks. 1) you can't use the SLIN directly on a non-intel machine - you may have to byte swap it first (took me a while to work out why I just got pulse modulated static on my NSLU2 home asterisk! (armv5teb) ) 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. I instantly felt like a child again, being told slowly and clearly what to do. The only reason I mention this is so people don't assume that these new recordings will always be the preferred offering to systems installed in the UK, it will depend on the image that a company wishes to present. Tim. Mark -- Mark Phillips [EMAIL PROTECTED] ___ --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 Tim Panton [EMAIL PROTECTED] ___ --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] British English voice files are ready fordownload
I downloaded the SLIN and I have a couple of remarks. 1) you can't use the SLIN directly on a non-intel machine - you may have to byte swap it first (took me a while to work out why I just got pulse modulated static on my NSLU2 home asterisk! (armv5teb) ) Can you explain in a bit more detail please, or post a link to some info on this topic? What procedure did you undertake to byte swap it? I'm using the voice prompts on my 2 datacentre machines, and my asterisk box at home (all intel), so I haven't come across this issue. Thanks for the warning - I was going to upload the files to a few clients' boxes this afternoon (which are all AMD boxes). 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. As you rightly say, it's a personal opinion. Personally, I rather like hearing a voice that talks neutrally rather than with a heavy accent which might be difficult to understand for people not of that locality, or speech that misses out sounds in words (t is normally an early victim). The only reason I mention this is so people don't assume that these new recordings will always be the preferred offering to systems installed in the UK, it will depend on the image that a company wishes to present. Are there any other royalty-free sets of recordings aimed at the UK market? I've seen a couple of others, one female, one male, but I don't think either is a complete set, which means from time to time you still get Allison (USA) chirping in with the odd sound or two. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] British English voice files are ready for download
If you need g729 and g723 format, let me know and i could convert it to you. Tim Panton escribió: On 19 May 2006, at 17:05, Mark Phillips wrote: Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} It is great that there is an extra (British) choice for asterisk sounds. I downloaded the SLIN and I have a couple of remarks. 1) you can't use the SLIN directly on a non-intel machine - you may have to byte swap it first (took me a while to work out why I just got pulse modulated static on my NSLU2 home asterisk! (armv5teb) ) 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. I instantly felt like a child again, being told slowly and clearly what to do. The only reason I mention this is so people don't assume that these new recordings will always be the preferred offering to systems installed in the UK, it will depend on the image that a company wishes to present. Tim. Mark -- Mark Phillips [EMAIL PROTECTED] ___ --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 Tim Panton [EMAIL PROTECTED] ___ --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] Upgrade 7960 from SCCP 3.0 to SIP 7.5
On Sun, 2006-05-21 at 14:28 +0200, Olivier Krief wrote: Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol to which you are converting (load_information for SCCP or image_version for SIP). 3. Remove any protocol configuration files that are not used for the specified protocol. Firmware versions are P003F300 (Application Load ID) and PC030300 (Boot Load ID). In TFTP root directory, the following files are present: OS79XX.TXT P003-07-5-00.bin P003-07-5-00.sbn P0S3-07-5-00.bin P0S3-07-5-00.sbn P0S3-07-5-00.loads SIPMAC Addres.cnf SIPDefault.cnf On boot, I can see in my TFTP logs the phone is served an OS79XX.TXT file which now holds P0S3-07-5-00 content. From TFTP logs, I can see my phone is then asking for P0S3-07-.bin file which doesn't exist in my TFTP directory. Next it asked for SEPMAC Addres.cnf and SEPDefault.cnf. Both files don't exist but SIPMAC Addres.cnf and SIPDefault.cnf do exist. In SIPDefault.cnf image_version=P0S3-07-5-00 is included. . When I change P0S3-07-5-00 to P003-07-5-00 in OS79XX.TXT file, it directly asked SEPMAC Addres.cnf and SEPDefault.cnf failing to ask for any .bin file. My first question is : What should be written is OS79XX.TXT if I want to upgrade to SIP ? P0S3-07-5-00 ? P0S307500 (with a symbolic link to P0S3-07-5-00 in TFTP root directory) ? P003-07-5-00 ? You have to upgrade to a new version of SCCP or older version of SIP before the bootloader on the phone will be able to handle the newer firmware. In the same Cisco page you read the info is there - you can either use an older version of SIP first, or a newer version of SCCP.. Older SIP is probably easier - 6.3 is the newest you can use to then jump to 7.x and/or 8.x.. You will need to put this in SEPDefault.cnf (not SIPDefault) image_version:P0S3-07-5-00 (whatever version you grab) That will tell it to grab the SIP firmware if it is not using OS79XX.txt - I cant remember that far back if it is still used.. Doesn't hurt to have both though... -Greg ___ --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] Limit outgoing calls
Does anyone have an idea how to limit the number of outging calls on a sip trunk . limit=x only works for incoming calls. ___ --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] transfer outside of a call?
Juraj Bednar wrote: I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Since Flash Operator Panel can do it, it's obviously possible. FOP uses the Asterisk Manager Interface (AMI) to do these things. ___ --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] British English voice files are ready fordownload
On 21 May 2006, at 16:53, Chris Bagnall wrote: I downloaded the SLIN and I have a couple of remarks. 1) you can't use the SLIN directly on a non-intel machine - you may have to byte swap it first (took me a while to work out why I just got pulse modulated static on my NSLU2 home asterisk! (armv5teb) ) Can you explain in a bit more detail please, or post a link to some info on this topic? What procedure did you undertake to byte swap it? Oh, sorry. The Slin files contain the audio data in a 16 bit format, each sample takes 2 bytes, but the byte order is cpu dependent. My home asterisk is running on a linksys NSLU2 (Slug). The CPU in it is an ARM - and it has the reverse byte order from an x86 system. I used : dd conv=swab to swap the bytes. I'm using the voice prompts on my 2 datacentre machines, and my asterisk box at home (all intel), so I haven't come across this issue. Thanks for the warning - I was going to upload the files to a few clients' boxes this afternoon (which are all AMD boxes). You won't have a problem with the slin files on an AMD - or any x86 box. Tim. Tim Panton [EMAIL PROTECTED] ___ --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] Limit outgoing calls
Does anyone have an idea how to limit the number of outging calls on a
sip trunk . limit=x only works for incoming calls.
__
Hi,
in the context where you dial out from:
exten = _X.,1,Set(GROUP()=OUTBOUND_GROUP)
exten = _X.,2,GotoIf($[${GROUP_COUNT()} 30 ] ? 4)
exten = _X.,4,NoOp(This trunk has more than 30 calls)
thanks,
yusuf
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Re: [Asterisk-Users] WiFi VoIP Handsets..
At 08:37 AM 5/21/2006, you wrote: I know it supports additional Aastra handsets, up to 10 I believe. As for 3rd party, not sure about that... Sounds better than it is, unless something's changed you can have 10 handsets but only 2 or 4 active calls. Ira ___ --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] British English voice files are ready for download
Obviously a Radio 1 listener. 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. I instantly felt like a child again, being told slowly and clearly what to do. ___ --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] Upgrade 7960 from SCCP 3.0 to SIP 7.5
2006/5/21, Greg Oliver [EMAIL PROTECTED]: You have to upgrade to a new version of SCCP or older version of SIPbefore the bootloader on the phone will be able to handle the newerfirmware.In the same Cisco page you read the info is there - you can either use an older version of SIP first, or a newer version of SCCP..Older SIP is probably easier - 6.3 is the newest you can use to thenjump to 7.x and/or 8.x..You will need to put this in SEPDefault.cnf (not SIPDefault)image_version:P0S3-07-5-00(whatever version you grab)That will tell it to grab the SIP firmware if it is not using OS79XX.txt- I cant remember that far back if it is still used..Doesn't hurt to have both though...-GregIn TFTP root directory, I didn't copy P0S30100.bin file as it is wasn't clear for me from that Cisco page, that I could or not use P0S3-07-5-00 file instead. From what Greg mentioned, I understand that this file is needed to allow upgrade from SCCP 3.0 to SIP 1.0, and hopefully from SIP 1.0 to SIP 7.5.I will also rename configuration file from SIPxxx to SEPxxx. I will provide feedback to the list. Thanks for all, anyway. ___ --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] Which is the best fax-modem for testing ?
Hi,2006/5/17, Rich Adamson [EMAIL PROTECTED]: (*) By fax doesn't hangup, I mean though Asterisk server forward an incoming fax call to the right extension, it keeps on ringing the fax machine which never hangup. Maybe the flash signal is too weak I'm very confused by the above statement.What do you mean by it keeps on ringing and machine never hangup inthe same sentence?(No such thing as it keeps ringing and never hangup. Hangup occurs after answering, so if its ringing, it can't hangup.)What do you mean by flash signal is too weak? (There's no such thingas a weak flash. Sort of equivalent to saying a weak binary 1.) My statement was very confused and I really apologize for that.It should have been written that way :By fax doesn't hangup, I mean though Asterisk server forward anincoming fax call to the right extension, it keeps on ringing the fax machine which never answer. Maybe the flash signal is too weakBest regards ___ --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] Which is the best fax-modem for testing ?
2006/5/21, Steve Underwood [EMAIL PROTECTED]: Lee Howard wrote: Olivier Krief wrote: For example, it seems that Brother 8360P uses Super G3 mode. Is there a fax-modem offering such capability so that I could easily check if I still cannothangup when I enable or disable Super G3 mode ? MultiTech 5634-series and MainPine RockForce fax modems (Agere chipset) support SuperG3.You'd run these with HylaFAX, for example, and not Asterisk.It is worth pointing out that the V.34 modems have almost no chance ofachieving V.34 speeds if you go:PSTN-analogue line-asterisk-FXS port-modemif you goPSTN-digital line-asterisk-FXS port-modem performance will depend on the FXS port, and any internal timing issues.With a TDM400 card its fairly unlikely to work. With a channel bankconnected to a port on the same digital card that connects to the PSTN chances are high.The problem with the PSTN-analogue line-asterisk-FXS port-modem pathis signal degradation through the extra analogue-digital-analogue stepis too much for V.34. For FAX modems up to V.29 it is no problem. ForV.17 is tends to work if the port quality is good.SteveHi Steve,Which fax-modem would you pick to highlight this behaviour ?I mean :If you had to buy a single fax-modem to complement a laptop to demonstrate a TDM or ToIP system is V.34 or V.17-capable, which fax-modem would you choose ?You launch a shell-script from your laptop and it sends 5 or 6 faxes with the same content to a given destination (always the same one) at different speeds or protocols. Reading destination fax machine's reception report, you can rate each sending and tellwhat your System Under Test is capable of.Cheers ___ --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] WiFi VoIP Handsets..
Are any of these FCC licensed for use in the USA. DECT in the USA is VERY new. Michael Graves wrote: HmmmI have a 480i-CT. Does this mean that I might be able to add third party DECT handsets? Or just the matching Aastra handsets? Michael --Original Message Text--- *From:* Dovid Bender *Date:* Sat, 20 May 2006 12:54:54 -0700 (PDT) Cory, Do you have the Nokia E70 and or the E60 ? If not are you guys gona get it in anytime soon ? Dovid */Cory Andrews [EMAIL PROTECTED]/* wrote: The Aastra 480i-CT and Uniden UIP1868 are both SIP based and support remote, wireless handsets via DECT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Tuesday, May 16, 2006 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. James Harper wrote: I was looking for something like this a while back (actually, a wifi + gsm combo), and came to the conclusion that a dect + gsm phone would be a better option, except that they don't exist (much). Maybe a VoIP capable DECT base station would be a better option for you? These do exist. James Thanks for all the replies.. James, you probably have a good point, a DECT cordless with a VoIP base station would probably work better for the situation I need to cater for.. Any pointers to recommended DECT VoIP phones? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Which is the best fax-modem for testing ?
Olivier Krief wrote: 2006/5/21, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Lee Howard wrote: Olivier Krief wrote: For example, it seems that Brother 8360P uses Super G3 mode. Is there a fax-modem offering such capability so that I could easily check if I still cannot hangup when I enable or disable Super G3 mode ? MultiTech 5634-series and MainPine RockForce fax modems (Agere chipset) support SuperG3. You'd run these with HylaFAX, for example, and not Asterisk. It is worth pointing out that the V.34 modems have almost no chance of achieving V.34 speeds if you go: PSTN-analogue line-asterisk-FXS port-modem if you go PSTN-digital line-asterisk-FXS port-modem performance will depend on the FXS port, and any internal timing issues. With a TDM400 card its fairly unlikely to work. With a channel bank connected to a port on the same digital card that connects to the PSTN chances are high. The problem with the PSTN-analogue line-asterisk-FXS port-modem path is signal degradation through the extra analogue-digital-analogue step is too much for V.34. For FAX modems up to V.29 it is no problem. For V.17 is tends to work if the port quality is good. Steve Hi Steve, Which fax-modem would you pick to highlight this behaviour ? I mean : If you had to buy a single fax-modem to complement a laptop to demonstrate a TDM or ToIP system is V.34 or V.17-capable, which fax-modem would you choose ? You launch a shell-script from your laptop and it sends 5 or 6 faxes with the same content to a given destination (always the same one) at different speeds or protocols. Reading destination fax machine's reception report, you can rate each sending and tell what your System Under Test is capable of. I thought I had clearly said this was related to the nature of the path, and has little to do with the specific modem you use. Steve ___ --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] Net2phone on asterisk
Has anyone setup a n2p account into asterisk? ___ --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] Re: Configuring a TDM400P with one FXS port
mohamed kerbachi wrote: Hi, I have a TDM400P and it works, Send us the output of your dmesg command. Regards. --- M.Hockings [EMAIL PROTECTED] a écrit : In my attempt to setup a single FXS line I have been following the instructions for Telephony Card Drivers on the asteriskdocs.org site. I have managed to checkout, make and install the zaptel code and can load the zaptel module but when I attempt to load the wcfxs module it tells me: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wctdm On the card the single FXS module is in the position at the back of the TDM400P (i.e., closest to the connectors) In the /etc/zaptel.conf file I have put the following at the bottom of the file: fxoks=1 loadzone=us defaultzone=us I have also tried fxoks=4 with the same results other than that the channel number changes in the message. Any idea what I am configuring incorrectly ? Mike Hi Mohamed, if you can help guide me in the right direction I would be most appreciative. The output of dmesg is below. In the machine is a TDM400P with one FXS, a X101P FXO, an ethernet card and an old ISA modem. Mike Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 BIOS-provided physical RAM map: BIOS-e820: - 0009fc00 (usable) BIOS-e820: 0009fc00 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 17ffa000 (usable) BIOS-e820: 17ffa000 - 17ffe000 (ACPI data) BIOS-e820: 17ffe000 - 1800 (ACPI NVS) 0MB HIGHMEM available. 383MB LOWMEM available. Using x86 segment limits to approximate NX protection zapping low mappings. On node 0 totalpages: 98298 DMA zone: 4096 pages, LIFO batch:1 Normal zone: 94202 pages, LIFO batch:16 HighMem zone: 0 pages, LIFO batch:1 DMI 2.0 present. ACPI: RSDP (v000 IBM ) @ 0x000fe030 ACPI: RSDT (v001 IBMV66XA0x0001 Acer 0x) @ 0x17ffa000 ACPI: FADT (v001 IBMV66XA0x0001 Acer 0x) @ 0x17ffa028 ACPI: DSDT (v001 IBMV66XA 0x1000 MSFT 0x010a) @ 0x ACPI: BIOS age (1998) fails cutoff (2001), acpi=force is required to enable ACPI ACPI: Disabling ACPI support Built 1 zonelists Kernel command line: ro root=/dev/VolGroup00/LogVol00 quiet Initializing CPU#0 CPU 0 irqstacks, hard=c03e7000 soft=c03e6000 PID hash table entries: 2048 (order: 11, 32768 bytes) Detected 400.948 MHz processor. Using tsc for high-res timesource Console: colour VGA+ 80x25 Dentry cache hash table entries: 65536 (order: 6, 262144 bytes) Inode-cache hash table entries: 32768 (order: 5, 131072 bytes) Memory: 384696k/393192k available (2117k kernel code, 7884k reserved, 669k data, 144k init, 0k highmem) Calibrating delay using timer specific routine.. 803.04 BogoMIPS (lpj=401522) Security Scaffold v1.0.0 initialized SELinux: Initializing. SELinux: Starting in permissive mode There is already a security framework initialized, register_security failed. selinux_register_security: Registering secondary module capability Capability LSM initialized as secondary Mount-cache hash table entries: 512 (order: 0, 4096 bytes) CPU: After generic identify, caps: 0183f9ff CPU: After vendor identify, caps: 0183f9ff CPU: L1 I cache: 16K, L1 D cache: 16K CPU: L2 cache: 512K CPU: After all inits, caps:0183f1ff 0040 Intel machine check architecture supported. Intel machine check reporting enabled on CPU#0. CPU: Intel Pentium II (Deschutes) stepping 02 Enabling fast FPU save and restore... done. Checking 'hlt' instruction... OK. checking if image is initramfs... it is Freeing initrd memory: 983k freed NET: Registered protocol family 16 PCI: PCI BIOS revision 2.10 entry at 0xf0200, last bus=1 PCI: Using configuration type 1 mtrr: v2.0 (20020519) ACPI: Subsystem revision 20040816 ACPI: Interpreter disabled. Linux Plug and Play Support v0.97 (c) Adam Belay usbcore: registered new driver usbfs usbcore: registered new driver hub PCI: Probing PCI hardware PCI: Probing PCI hardware (bus 00) PCI: Using IRQ router PIIX/ICH [8086/7110] at :00:07.0 IBM machine detected. Enabling interrupts during APM calls. apm: BIOS version 1.2 Flags 0x03 (Driver version 1.16ac) audit: initializing netlink socket (disabled) audit(1148245664.147:1): initialized Total HugeTLB memory allocated, 0 VFS: Disk quotas dquot_6.5.1 Dquot-cache hash table entries: 1024 (order 0, 4096 bytes) SELinux: Registering netfilter hooks Initializing Cryptographic API ksign: Installing public key data Loading keyring - Added public key B4802E7A21D4FA03 - User ID: CentOS (Kernel Module GPG key) Limiting direct PCI/PCI transfers. pci_hotplug: PCI Hot Plug PCI Core version: 0.5 Real Time Clock Driver v1.12 Linux agpgart interface
