We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks
blackgecko wrote:>We just bougth a tdm2400p with all the modules for FXO, but we are >having some troubles with the card, cause it aparently is stripping >some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. > >We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug >Asterisk does NOT listen for dialtone before dialing. Many consider that a bug, or a design defect. Unfortunately no one who has the skills in coding to fix that sees it as an issue. Multiple w's may fix it if that is really the problem. Remember that w to wait before dialing ONLY works in DTMF. If you are forced or want to use pulse dialing, too bad. w doesn't work in that case. You may want to monitor a line while dialing out and see if that is really the problem, though I would think it should be there with the TDM400 as well. John Novack
I still have the problem, im using DTMF not pulses, and the problem isnt happening with the tdm400, does anyone has any clue?? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060423/064a8533/attachment.htm
Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycom's IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 =============== Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel => 1-4 channel => 5-8 signalling = fxo_ks channel => 17-20 channel => 21-24 Best Regards, Robson Ribeiro MSN: robrib2002@hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060921/e8610ed1/attachment.htm
Try turning off of echotraining. Sometimes it does more harm than good. Matthew Fredrickson On Sep 21, 2006, at 11:28 AM, Robson Ribeiro wrote:> > Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. > They are installed respectively on banks 1,2,5 and 6. The problem I am > having is that when I make a call using the ZAP channel, I can hear > perfectly but the person on the other end is hearing my voice with > lots of ticks. It would seem I am making this call over a very bad > bandwidth which is not the case since this is the PSTN. My > configuration files are below, I have the latest versions of Zaptel, > Libpri and Asterisk. I am using Polycom?s IP301 and IP430 Phones. I > would appreciate help since I have to put this in production on > Saturday. > ? > # Zaptel Configuration File > # > # This file is parsed by the Zaptel Configurator, ztcfg > # > # > > loadzone = us > defaultzone=us > fxsks=1-4 > fxsks=5-8 > fxoks=17-20 > fxoks=21-24 > > ===============> > Zapata.conf > > [channels] > language=en > context=default > ;switchtype=national > echocancel=64 > echocancelwhenbridged=no > echotraining=800 > toneduration=200 > busydetect=yes > signalling = fxs_ks > rxgain=5.0 > txgain=-10.0 > channel => 1-4 > channel => 5-8 > signalling = fxo_ks > channel => 17-20 > channel => 21-24 > ? > ? > Best Regards, > ? > Robson Ribeiro > MSN: robrib2002@hotmail.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
This is definately a t-1 slipping, check your timing on the spans, also ensure that you have replaced the t-1 cables. I see this most of the time caused by cables that were made by hand. Usually because a solid conductor was used with stranded wire or solid cat5 with stranded rj-45's. If possible use a premade patch cord made of stranded wire. The interupt would be my second step, although this usually makes the t completley unusable, especially using all four ports. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060921/462027d0/attachment.htm
I had a similar problem and the problem was the hardware echo can who was defect. Try removing the echo can hardware echo can and test the line. David _____ De : asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] De la part de Robson Ribeiro Envoy? : 21 septembre 2006 12:28 ? : asterisk-users@lists.digium.com Objet : [asterisk-users] TDM2400P Importance : Haute Hi all, I have a TDM2400P w/ echo cancellation, 8 FXSs and 8 FXOs. They are installed respectively on banks 1,2,5 and 6. The problem I am having is that when I make a call using the ZAP channel, I can hear perfectly but the person on the other end is hearing my voice with lots of ticks. It would seem I am making this call over a very bad bandwidth which is not the case since this is the PSTN. My configuration files are below, I have the latest versions of Zaptel, Libpri and Asterisk. I am using Polycom?s IP301 and IP430 Phones. I would appreciate help since I have to put this in production on Saturday. # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = us defaultzone=us fxsks=1-4 fxsks=5-8 fxoks=17-20 fxoks=21-24 =============== Zapata.conf [channels] language=en context=default ;switchtype=national echocancel=64 echocancelwhenbridged=no echotraining=800 toneduration=200 busydetect=yes signalling = fxs_ks rxgain=5.0 txgain=-10.0 channel => 1-4 channel => 5-8 signalling = fxo_ks channel => 17-20 channel => 21-24 Best Regards, Robson Ribeiro MSN: robrib2002@hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060921/6c11a945/attachment.htm