I had problems with a system I was trying to bring up using a couple older a104d cards we had lying around. Neither card would pass audio. I worked with one Sangoma tech for a couple hours while he tried various things. The second tech I worked with got on the system and updated the firmware for the cards. When I tried to show him the problem things worked. I said "you did something as this did not work an hour ago". He told me the first think he does when troubleshooting is to update the firmware to the current version. A lesson I have now learned. I do that with software but rarely remember to look for firmware updates. Take a look at wiki.sangoma.com and it lets you know current firmware versions as well as how to update if you are not running the current version. -- Jim Dickenson mailto:dicken...@cfmc.com
CfMC http://www.cfmc.com/ On Apr 25, 2011, at 4:41 PM, Edwin Lam wrote: > i think i have similar problem after upgraded from 1.4.x to 1.6.2.17. > (originally upgraded to 1.8.3.2 unfortunately there were other more > pressing problems that forced me to downgraded it to 1.6.2.17) > i have a wanpipe device with 2 channels uses PRI signalling to PSTN & > the other 2 uses FXO signalling (connect to Rhino FXS channel bank). > the PRI part works fine but the FXO channels are having DTMF digits > skipped. i'm still trying to find out what's wrong with it. > > On 4/23/11 8:48 AM, David wrote: >> Hello, >> I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple >> problems with DTMF. >> I have two machines, we'll call them asterisk and asterisk-pri. Asterisk >> does IVR >> and asterisk-pri has a PRI card in it and connects to the PSTN. The two >> servers >> communicate via SIP with RFC2833. >> I setup logger.conf on both machines to display DTMF to the console. Both are >> built from source. >> Asterisk : spandsp, dahdi, asterisk. >> Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe >> I eliminated AGI, hard phones, network et al by setting up this extension : >> exten => 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 >> <mailto:SIP/114186939...@pri1.omnity.net,30,D(132412983>#)) >> in default. >> The only other non default setting is in sip.conf I added a outboundproxy ( >> which >> does NOT do RTP, only SIP ). >> I called asterisk from my hard phone ( gxp2000 ) by dialing 22. >> I see the console DTMF messages indicating the DTMF was sent or received. ( I >> forgot to keep this output ). >> I than watch the console DTMF output on asterisk-pri and it showed about >> half the >> DTMFs. The pager that was called showed the DTMFs that appeared on the >> asterisk-pri console. >> So somewhere between the two machines, the DTMFs have disappeared. So I ran >> TCPDump on asterisk and saw that close to half of the DTMF events were never >> sent. >> tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap >> I imported the file into wireshark on my local machine and confirmed that >> the dump >> almost matches what I saw on asterisk-pri. >> So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. >> I compared the packet scan to what I saw on asterisk-pri and noticed that >> between >> 1 and 3 dtmfs were missing. >> Problem 2 : Asterisk-pri loses some received DTMFs. >> I also noticed that some of the DTMFs coming out of asterisk had the wrong >> Event >> Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 >> seconds ) but I only pressed the button for like 1/3 of a second. >> What I do not understand is that I in my final test last night was using >> asterisk >> 1.6 current with centos ( os that asterisk is developed on from my >> understanding ) >> with all default settings ( excluding logger.conf, dialplan and >> outboundproxy ) >> and I am having problems with the DTMF. >> Both servers were installed with CentOS 5.5 and were updated last night, >> after >> which I reinstalled asterisk. This did not resolve the issue. >> I am at wit's end and do not know where to go from here. I would really >> appreciate >> it if someone could give me some pointers on where to go next, what >> additionnal >> debugging steps I should perform. I would also really appreciate if someone >> could >> propose a solution. >> Please help! >> David >> Never give up, never surrender > > -- > Edwin Lam <edwin....@officegeneral.com> > Systems Engineer, OfficeWyze, Inc. > Ph: +1 415 439 4988 Fax: +1 415 283 3370 > http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 > > > -- > _____________________________________________________________________ > -- 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
