Re: [asterisk-users] Help choosing the right card
Eric, I sure did. It is active for the last 18 months ever since I started having this problem which coincided with my switching to Asterisk 1.8 from 1.6.2.x where I never ever had any of the DAHDI and / or Asterisk-DAHDI problems I described before. -Vladimir On 6/17/2012 12:48 AM, Eric Wieling wrote: You have verified this by using the Asterisk's DTMF debug option? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 9:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Eric, Thank you for the suggestion. In fact the problem is with FSX channel which fails to catch some DTMF tones from a phone which places an outgoing call. Shaun's theory was a delay related to swapping. -Vladimir On 6/16/2012 7:40 PM, Eric Wieling wrote: I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to increase the duration of transmitted DTMF on your DAHDI channels. If that fixes it, try lowering it. I find 80 usually works with even the worst IVRs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On Sun, Jun 17, 2012 at 04:39:55PM +0200, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. This is what the patch on DAHLIN-241 [1] is intended to do (only if Asterisk is run in the real-time priority class) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On Sun, 17 Jun 2012, Shaun Ruffell wrote: What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. Any chance there is a USB or 'green' disk drive going to sleep anywhere? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Steve, The systems I tested on are all old Dell Dimension systems with plain old PATA. I disabled all power saving features in the BIOS. -Vladimir On 6/17/2012 12:57 PM, Steve Edwards wrote: On Sun, 17 Jun 2012, Shaun Ruffell wrote: What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. Any chance there is a USB or 'green' disk drive going to sleep anywhere? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Benny, Thank you for clarification. I did not know executable pages would not swap. Shaun wrote a patch with mlockall() and it crashed my system badly. Is there a way to lock just a specific module in memory vs. the whole Asterisk application? -Vladimir On 6/17/2012 9:39 AM, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On 6/17/2012 12:06 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 04:39:55PM +0200, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. This is what the patch on DAHLIN-241 [1] is intended to do (only if Asterisk is run in the real-time priority class) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB -Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On Sun, Jun 17, 2012 at 03:43:35PM -0500, Vladimir Mikhelson wrote: On 6/17/2012 12:06 PM, Shaun Ruffell wrote: I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB You are correct. You will not be able to lock all the memory and one of the allocations will fail if you're autoloading all modules. So if you want to avoid delays incurred when the system needs to page in code pages on events, you will either need to add more memory or limit the modules that are loaded. For example, when I disable autoloading, and only load a few modules needed for a basic system only 153M is needed: # asterisk -rx 'module show' Module Description Use Count pbx_config.so Text Extension Configuration 0 res_timing_dahdi.soDAHDI Timing Interface 0 chan_dahdi.so DAHDI Telephony Driver w/PRI 0 chan_sip.soSession Initiation Protocol (SIP)0 app_dial.soDialing Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_originate.so Originate call 0 app_meetme.so MeetMe conference bridge 0 codec_ulaw.so mu-Law Coder/Decoder 0 format_sln.so Raw Signed Linear Audio support (SLN)0 format_sln16.soRaw Signed Linear 16KHz Audio support (S 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 codec_alaw.so A-law Coder/Decoder 0 func_callerid.so Party ID related dialplan functions (Cal 0 func_version.soGet Asterisk Version/Build Info 0 16 modules loaded # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:152972 kB -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On 6/17/2012 5:56 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 03:43:35PM -0500, Vladimir Mikhelson wrote: On 6/17/2012 12:06 PM, Shaun Ruffell wrote: I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB You are correct. You will not be able to lock all the memory and one of the allocations will fail if you're autoloading all modules. So if you want to avoid delays incurred when the system needs to page in code pages on events, you will either need to add more memory or limit the modules that are loaded. For example, when I disable autoloading, and only load a few modules needed for a basic system only 153M is needed: # asterisk -rx 'module show' Module Description Use Count pbx_config.so Text Extension Configuration 0 res_timing_dahdi.soDAHDI Timing Interface 0 chan_dahdi.so DAHDI Telephony Driver w/PRI 0 chan_sip.soSession Initiation Protocol (SIP)0 app_dial.soDialing Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_originate.so Originate call 0 app_meetme.so MeetMe conference bridge 0 codec_ulaw.so mu-Law Coder/Decoder 0 format_sln.so Raw Signed Linear Audio support (SLN)0 format_sln16.soRaw Signed Linear 16KHz Audio support (S 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 codec_alaw.so A-law Coder/Decoder 0 func_callerid.so Party ID related dialplan functions (Cal 0 func_version.soGet Asterisk Version/Build Info 0 16 modules loaded # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:152972 kB Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? -Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On Sun, Jun 17, 2012 at 06:14:07PM -0500, Vladimir Mikhelson wrote: Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? It is possible but not without more work. Asterisk would need to parse the output of the memory map in /proc/pid/maps and figure out where the modules are mapped into the current process' address space and then lock only those pages. Also, this would require knowing exactly which modules are needed at first. Since Asterisk really should be run in a soft real-time fashion, I still believe it's preferrable to figure out which modules are needed and then making sure all those pages can stay resident in memory. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On 6/17/2012 6:21 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 06:14:07PM -0500, Vladimir Mikhelson wrote: Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? It is possible but not without more work. Asterisk would need to parse the output of the memory map in /proc/pid/maps and figure out where the modules are mapped into the current process' address space and then lock only those pages. Also, this would require knowing exactly which modules are needed at first. Since Asterisk really should be run in a soft real-time fashion, I still believe it's preferrable to figure out which modules are needed and then making sure all those pages can stay resident in memory. Shaun, Thank you for the reply. I would suggest to move this conversation to JIRA as we digressed from the original topic a lot. https://issues.asterisk.org/jira/browse/DAHLIN-241?focusedCommentId=193920page=com.atlassian.jira.plugin.system.issuetabpanels%3Acomment-tabpanel#comment-193920 Thank you, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Although it was better to ask it in Asterisk commercial list but you have different options like Digium, Sangoma or Openvox. TDM410P is the PCI one from Digium which suits your description. Just remember to buy two trunk (FXO) modules too and if you are looking for a best sound qulity get hardware echo canceller too. I just didnt get why your are going to set 5 ext on each IP Phone!!! -- M. Shokuie Nia On Sat, Jun 16, 2012 at 4:34 PM, Amit Patel pistolfir...@gmail.com wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Poor grammar on my part, What I meant was to assign one ext to each IP Phone, my initial setup consists of 5 phones. If all things work out as planned and after better understanding I wish to support upto around 70 IP phones. I do plan to get echo canceller too ;) Thanks for the reply. On Sat, Jun 16, 2012 at 8:08 PM, M Shokuie sena...@gmail.com wrote: Although it was better to ask it in Asterisk commercial list but you have different options like Digium, Sangoma or Openvox. TDM410P is the PCI one from Digium which suits your description. Just remember to buy two trunk (FXO) modules too and if you are looking for a best sound qulity get hardware echo canceller too. I just didnt get why your are going to set 5 ext on each IP Phone!!! -- M. Shokuie Nia On Sat, Jun 16, 2012 at 4:34 PM, Amit Patel pistolfir...@gmail.comwrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Amit, Make sure you have an option to return Digium TDM410P if it does not work for you. In my experience Digium TDM410P produce substantial background noise on certain Dell computers. Generic TDM400 do not have this issue. On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. -Vladimir On 6/16/2012 7:04 AM, Amit Patel wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
In my experience when you have intermittent problems with incoming caller ID, FXS -- with DTMF detection you have to adjust your rxgain and/or txgain. I am NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and solvable and not related to the card itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Amit, Make sure you have an option to return Digium TDM410P if it does not work for you. In my experience Digium TDM410P produce substantial background noise on certain Dell computers. Generic TDM400 do not have this issue. On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. -Vladimir On 6/16/2012 7:04 AM, Amit Patel wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote: On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. Why do you believe it is a deficiency in DAHDI? The only issue that I know of related to DTMF detection is DAHLIN-241 [1], which you and I are currently trying to definitely establish is related to the time it takes to page Asterisk pages back into memory, which isn't really a DAHDI issue. [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 Also, the only recent CID issue I'm aware of which was related to DAHDI was introduced in 2.6.0 and fixed in 2.6.1 [2]. This issue was that the ring event was reported late to Asterisk, so Asterisk would bump the gains in the middle of the spill. If the line quality was marginal the callerid would be missed. [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10473 Are there other issues with CID detection that you are currently having? Generally CID issues aren't necesarily DAHDI related since generation and detection is done in Asterisk and not DAHDI, like DTMF detection (DTMF generation can be done in DAHDI...). Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Eric, I wish it were that simple as you described. rxgain, txgain, cidrxgain, Digium support, Jira ticket, hours of troubleshooting, hardware swaps, external power source, etc. -- all these are the options I tried so far. BTW, I did state that these specific issues are not limited to Digium TDM410P cards. -Vladimir On 6/16/2012 5:15 PM, Eric Wieling wrote: In my experience when you have intermittent problems with incoming caller ID, FXS -- with DTMF detection you have to adjust your rxgain and/or txgain. I am NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and solvable and not related to the card itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Amit, Make sure you have an option to return Digium TDM410P if it does not work for you. In my experience Digium TDM410P produce substantial background noise on certain Dell computers. Generic TDM400 do not have this issue. On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. -Vladimir On 6/16/2012 7:04 AM, Amit Patel wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On 6/16/2012 5:38 PM, Shaun Ruffell wrote: On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote: On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. Why do you believe it is a deficiency in DAHDI? The only issue that I know of related to DTMF detection is DAHLIN-241 [1], which you and I are currently trying to definitely establish is related to the time it takes to page Asterisk pages back into memory, which isn't really a DAHDI issue. [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Also, the only recent CID issue I'm aware of which was related to DAHDI was introduced in 2.6.0 and fixed in 2.6.1 [2]. This issue was that the ring event was reported late to Asterisk, so Asterisk would bump the gains in the middle of the spill. If the line quality was marginal the callerid would be missed. [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10473 Are there other issues with CID detection that you are currently having? Generally CID issues aren't necesarily DAHDI related since generation and detection is done in Asterisk and not DAHDI, like DTMF detection (DTMF generation can be done in DAHDI...). I am not sure whether it is DAHDI or Asterisk specifically, well may be one or both. All I know is I opened support ticket with Digium, tried all their recommendations, and I am still experiencing the same CID detection intermittency. Interestingly, in my case 2.6.0 was an improvement, whereas 2.6.1 returned everything back to where it was before. I still did not find time to file on JIRA, but I eventually will. Here are the issues (not detailed, from memory): 1. Incoming calls CID intermittency 2. Lack of a dropped call detection where 45 seconds long voice mails with CO generated bipping are left and the line is captured for up to 3 minutes after a caller dropped a call when the IVR was active 3. Intermittent outgoing DTMF drop 4. CID flags are not being passed into dial plan for analysis, so we cannot distinguish between private caller, no CID name, etc. Cheers, Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Eric, I wish it were that simple as you described. rxgain, txgain, cidrxgain, Digium support, Jira ticket, hours of troubleshooting, hardware swaps, external power source, etc. -- all these are the options I tried so far. BTW, I did state that these specific issues are not limited to Digium TDM410P cards. -Vladimir On 6/16/2012 5:15 PM, Eric Wieling wrote: In my experience when you have intermittent problems with incoming caller ID, FXS -- with DTMF detection you have to adjust your rxgain and/or txgain. I am NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and solvable and not related to the card itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Amit, Make sure you have an option to return Digium TDM410P if it does not work for you. In my experience Digium TDM410P produce substantial background noise on certain Dell computers. Generic TDM400 do not have this issue. On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. -Vladimir On 6/16/2012 7:04 AM, Amit Patel wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
On 6/16/2012 5:38 PM, Shaun Ruffell wrote: On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote: On top of that FXO channels exhibit intermittent problems with incoming caller ID, FXS -- with DTMF detection. These two problems manifest themselves with both Digium and generic cards. It looks like it is just a deficiency in the current DAHDI implementation. Why do you believe it is a deficiency in DAHDI? The only issue that I know of related to DTMF detection is DAHLIN-241 [1], which you and I are currently trying to definitely establish is related to the time it takes to page Asterisk pages back into memory, which isn't really a DAHDI issue. [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Also, the only recent CID issue I'm aware of which was related to DAHDI was introduced in 2.6.0 and fixed in 2.6.1 [2]. This issue was that the ring event was reported late to Asterisk, so Asterisk would bump the gains in the middle of the spill. If the line quality was marginal the callerid would be missed. [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10473 Are there other issues with CID detection that you are currently having? Generally CID issues aren't necesarily DAHDI related since generation and detection is done in Asterisk and not DAHDI, like DTMF detection (DTMF generation can be done in DAHDI...). I am not sure whether it is DAHDI or Asterisk specifically, well may be one or both. All I know is I opened support ticket with Digium, tried all their recommendations, and I am still experiencing the same CID detection intermittency. Interestingly, in my case 2.6.0 was an improvement, whereas 2.6.1 returned everything back to where it was before. I still did not find time to file on JIRA, but I eventually will. Here are the issues (not detailed, from memory): 1. Incoming calls CID intermittency 2. Lack of a dropped call detection where 45 seconds long voice mails with CO generated bipping are left and the line is captured for up to 3 minutes after a caller dropped a call when the IVR was active 3. Intermittent outgoing DTMF drop 4. CID flags are not being passed into dial plan for analysis, so we cannot distinguish between private caller, no CID name, etc. Cheers, Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to increase the duration of transmitted DTMF on your DAHDI channels. If that fixes it, try lowering it. I find 80 usually works with even the worst IVRs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
Eric, Thank you for the suggestion. In fact the problem is with FSX channel which fails to catch some DTMF tones from a phone which places an outgoing call. Shaun's theory was a delay related to swapping. -Vladimir On 6/16/2012 7:40 PM, Eric Wieling wrote: I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to increase the duration of transmitted DTMF on your DAHDI channels. If that fixes it, try lowering it. I find 80 usually works with even the worst IVRs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help choosing the right card
You have verified this by using the Asterisk's DTMF debug option? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 9:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Eric, Thank you for the suggestion. In fact the problem is with FSX channel which fails to catch some DTMF tones from a phone which places an outgoing call. Shaun's theory was a delay related to swapping. -Vladimir On 6/16/2012 7:40 PM, Eric Wieling wrote: I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to increase the duration of transmitted DTMF on your DAHDI channels. If that fixes it, try lowering it. I find 80 usually works with even the worst IVRs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
