Re: [asterisk-users] dahdi/DTMF problem
On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote: > incoming calls > through the FXO line are dropped as soon as there is a button press. > The error logged is: > > [Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2 > format > [Aug 23 18:15:39] WARNING[6532] file.c: Failed to write frame > > > Which looks like this bug: > > https://issues.asterisk.org/view.php?id=15129 I didn't solve this, but I worked around it. I eventually gave up and installed the "asterisk14" 1.4.26 packages from ATrpms. This version I was able to get working with Dahdi. I'll keep my eye on the bug report to see if this ever gets fixed, then I might try to upgrade to 1.6. But I have no urgent need to do so, so I am happy to wait a while and at least I can finally retire the old system. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi/DTMF problem
On Mon, Sep 7, 2009 at 7:50 PM, Greg Woods wrote: > On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote: > > incoming calls > > through the FXO line are dropped as soon as there is a button press. > > The error logged is: > > > > [Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2 > > format > > [Aug 23 18:15:39] WARNING[6532] file.c: Failed to write frame > > > > > > Which looks like this bug: > > > > https://issues.asterisk.org/view.php?id=15129 > > I didn't solve this, but I worked around it. I eventually gave up and > installed the "asterisk14" 1.4.26 packages from ATrpms. This version I > was able to get working with Dahdi. > > I'll keep my eye on the bug report to see if this ever gets fixed, then > I might try to upgrade to 1.6. But I have no urgent need to do so, so I > am happy to wait a while and at least I can finally retire the old > system. > > --Greg > I hope that you'll add yourself as a watcher or comment on the issue so that once somebody gets around to looking at it, you'll be notified and can assist. Jeff Peeler Digium, Inc. | Software 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI DTMF problem?
- Original Message - > I have an Asterisk server configured to run as voicemail with a T1 > and SMDI. > It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great > for a few > years. I am configuring a new server with Asterisk 1.8.13 (dahdi > 2.6.1) on > Centos 5.8 > > The problem I am having appears to be related to DTMF detection. When > the > test phone number is called (2704083000) Asterisk only receives a > portion of > the dialed number. It varies as to what numbers are detected. > Sometimes it > sees a single digit, sometimes 3 or 4 of the digits of the dialed > number. > > When I compare this to the old server the debug below is similar but > there > isn't any mention of the "sig_analog.c" lines shown below. > > I am told the T1's on the old server and the new server are > configured the > same. I can make outgoing calls on the T1 from Asterisk. > > Can someone give me a clue as to what could be causing this? > What kind of hardware are you using to interface with the T1? Is there any chance a HWEC or other DSP is interfering with DTMF? --Tim -- _ -- 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] DAHDI DTMF problem?
Sorry, forgot to include that. It has a Digium Wildcard TE122 I've asked Digium about the card below. They say the -1 in the bipolar and CRC errors is ok. They don't change. Description AlarmsIRQbpviol CRCFra Codi Options LBO Wildcard TE122 Card 0 OK1 -1 -1 ESF B8ZS0 db (CSU)/0-133 feet (DSX-1) Bill Dunn - Original Message - From: Tim Nelson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 11:01 AM Subject: Re: [asterisk-users] DAHDI DTMF problem? - Original Message - I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on Centos 5.8 The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number. When I compare this to the old server the debug below is similar but there isn't any mention of the "sig_analog.c" lines shown below. I am told the T1's on the old server and the new server are configured the same. I can make outgoing calls on the T1 from Asterisk. Can someone give me a clue as to what could be causing this? What kind of hardware are you using to interface with the T1? Is there any chance a HWEC or other DSP is interfering with DTMF? --Tim -- _ -- 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] DAHDI DTMF problem?
- Original Message - > > It has a Digium Wildcard TE122 > If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. --Tim -- _ -- 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] DAHDI DTMF problem?
On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote: > - Original Message - > > > > It has a Digium Wildcard TE122 > > If it has an onboard echo canceler, try disabling it and retrying. > Just a shot in the dark, going from my experience with other cards > and same symptoms. If the card is new(ish) I would think Digium > could provide support to you for determining the DTMF problems. Bill, To repeat what Tim said, if you're eligible I would recommend contacting Digium's support department. There are many variables with a new install and Digium support would be happy to help you troubleshoot. Some other things to try in order to isolate the drivers / hardware change from the Asterisk change. You could: a) Install the exact same versions of Asterisk / DAHDI that you used previously on the new server with Centos 5.8. b) Run a local pattern test to verify the host <-> card communication is valid. If you have problems here there may be a framebuffer configured or a disk controller running in combined mode preventing the cards interrupt handler from running in a timely fashion. c) Use dahdi_monitor to record the audio on the channel you're testing with and open it up with audacity and verify that the DTMF looks correct. If it does, then most likely there is a problem with chan_dahdi reading the audio from the drivers quickly enough. d) Ensure that you are only loading the Asterisk modules you need in case you're running on a system with limited memory and the asterisk process is dropping audio while paging in code. (See DAHLIN-241 [1]) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 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] DAHDI DTMF problem?
On Fri, Jul 06, 2012 at 11:07:26AM -0500, Bill Dunn - VCI Internet Services wrote: > > It has a Digium Wildcard TE122 > > I've asked Digium about the card below. They say the -1 in the > bipolar and CRC errors is ok. They don't change. > > Description AlarmsIRQbpviol CRC > Fra Codi Options LBO > Wildcard TE122 Card 0 OK1 -1 -1 > ESF B8ZS0 db (CSU)/0-133 feet (DSX-1) Some background on the -1s for the bipolar and CRC errors: The driver for the TE122 returns -1's now for the error counters to flag that the driver does not actually collect them versus returning 0 which was leading users to believe no errors existed. This change was made in r10212 "wcte12xp: Set uncollected performance counters to -1" [1]. [1] http://svnview.digium.com/svn/dahdi?view=revision&revision=10212 -- 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] DAHDI DTMF problem?
Bill Dunn - VCI Internet Services wrote: > I have an Asterisk server configured to run as voicemail with a T1 and > SMDI. > It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a > few > years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on > Centos 5.8 > > The problem I am having appears to be related to DTMF detection. When the > test phone number is called (2704083000) Asterisk only receives a portion > of > the dialed number. It varies as to what numbers are detected. Sometimes it > sees a single digit, sometimes 3 or 4 of the digits of the dialed number. > > When I compare this to the old server the debug below is similar but there > isn't any mention of the "sig_analog.c" lines shown below. > > I am told the T1's on the old server and the new server are configured the > same. I can make outgoing calls on the T1 from Asterisk. > > Can someone give me a clue as to what could be causing this? > > > Bill Dunn > Try setting: relaxdtmf=yes We used to have that same problem on most of our servers. Setting relaxdtmf to yes solved the problem for us. -- Ron Bergin Network Operations Administrator Fry's Electronics, Inc. -- _ -- 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] DAHDI DTMF problem?
Thanks Ron. I have had my chan_dahdi.conf file set as follows with the same result. [trunkgroups] [channels] switchtype=national usecallerid=yes callerid=asreceived cidsignalling=smdi echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 usesmdi=yes smdiport=/dev/ttyS0 signalling = em_w immediate = no group = 1 channel => 1-3 Bill Dunn - Original Message - From: Ron Bergin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 12:34 PM Subject: Re: [asterisk-users] DAHDI DTMF problem? Bill Dunn - VCI Internet Services wrote: I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on Centos 5.8 The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number. When I compare this to the old server the debug below is similar but there isn't any mention of the "sig_analog.c" lines shown below. I am told the T1's on the old server and the new server are configured the same. I can make outgoing calls on the T1 from Asterisk. Can someone give me a clue as to what could be causing this? Bill Dunn Try setting: relaxdtmf=yes We used to have that same problem on most of our servers. Setting relaxdtmf to yes solved the problem for us. -- Ron Bergin Network Operations Administrator Fry's Electronics, Inc. -- _ -- 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] DAHDI DTMF problem?
I used the dahdi_monitor to record the audio on the T1 channel of the working server and the new server. The audio stream of the working server allowed me to hear the audio I heard over the phone call plus the DTMF at the very beginning. The audio of the new server was completely messed up. I could barely make out the where the DTMF and the audio were in the file. If I didn't have a working sample I wouldn't know what it was. And, the beginning of the new server sample always contains a hum or tone in it whereas the working server does not have that. What does this tell me? Bill Dunn - Original Message - From: Shaun Ruffell To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 06, 2012 12:16 PM Subject: Re: [asterisk-users] DAHDI DTMF problem? On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote: - Original Message - > > It has a Digium Wildcard TE122 If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. Bill, To repeat what Tim said, if you're eligible I would recommend contacting Digium's support department. There are many variables with a new install and Digium support would be happy to help you troubleshoot. Some other things to try in order to isolate the drivers / hardware change from the Asterisk change. You could: a) Install the exact same versions of Asterisk / DAHDI that you used previously on the new server with Centos 5.8. b) Run a local pattern test to verify the host <-> card communication is valid. If you have problems here there may be a framebuffer configured or a disk controller running in combined mode preventing the cards interrupt handler from running in a timely fashion. c) Use dahdi_monitor to record the audio on the channel you're testing with and open it up with audacity and verify that the DTMF looks correct. If it does, then most likely there is a problem with chan_dahdi reading the audio from the drivers quickly enough. d) Ensure that you are only loading the Asterisk modules you need in case you're running on a system with limited memory and the asterisk process is dropping audio while paging in code. (See DAHLIN-241 [1]) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 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 -- _ -- 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] DAHDI DTMF problem?
Bill Dunn - VCI Internet Services wrote: > Thanks Ron. I have had my chan_dahdi.conf file set as follows with the > same > result. > > [trunkgroups] > [channels] > switchtype=national > usecallerid=yes > callerid=asreceived > cidsignalling=smdi > echocancel=yes > echocancelwhenbridged=yes > relaxdtmf=yes > rxgain=0.0 > txgain=0.0 > usesmdi=yes > smdiport=/dev/ttyS0 > signalling = em_w > immediate = no > group = 1 > channel => 1-3 > > > > Bill Dunn > > > > - Original Message - > From: Ron Bergin > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Friday, July 06, 2012 12:34 PM > Subject: Re: [asterisk-users] DAHDI DTMF problem? > > > Try setting: > relaxdtmf=yes > > We used to have that same problem on most of our servers. Setting > relaxdtmf to yes solved the problem for us. > Are you using SIP? If so, relaxdtmf can also be set in sip.conf as well as a dtmfmode setting that you can adjust. What type of phones are you using? In our case, part of this problem was due our low end 2.4ghz cordless phones. -- Ron Bergin Network Operations Administrator Fry's Electronics (408) 487-4600 -- _ -- 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] DAHDI DTMF problem?
On Fri, Jul 06, 2012 at 01:28:01PM -0500, Bill Dunn - VCI Internet Services wrote: > > I used the dahdi_monitor to record the audio on the T1 channel of the > working server and the new server. The audio stream of the working server > allowed me to hear the audio I heard over the phone call plus the DTMF at > the very beginning. The audio of the new server was completely messed up. I > could barely make out the where the DTMF and the audio were in the file. If > I didn't have a working sample I wouldn't know what it was. And, the > beginning of the new server sample always contains a hum or tone in it > whereas the working server does not have that. > > What does this tell me? Bill, to close out this thread: After logging into your system, I'm nearly certain you'll need to contact Digium technical support for assistance if moving the card into a different system doesn't change the results. I ran the same commands I ran on your server on a test server of mine and this is what I would have expected on your server: $ cat system.em.conf span=1,0,0,esf,b8zs e&m=1-24 loadzone= us defaultzone = us $ modprobe --first-time wcte12xp vpmsupport=0 latency=6 debug=1 $ dahdi_cfg -c system.em.conf $ dahdi_maint -s 1 --loopback localhost Span 1: local host loopback ON $ patlooptest 1 -t 10 && patlooptest 1 -t 10 Using Timeout of 10 Seconds Going for it... Timeout achieved Ending Program Test ran 33 loops of 2039 bytes/loop with 0 errors Using Timeout of 10 Seconds Going for it... Timeout achieved Ending Program Test ran 34 loops of 2039 bytes/loop with 0 errors On your server this same sequence would produce repeated patloop errors even though there were not any indications in dmesg of latency bumps, which would be case if something else was preventing the wcte12xp interrupt service routine from running. The patlooptest errors would explain the bad audio you recorded with dahdi_monitor and the DTMF detection problems. 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
