[Asterisk-Users] Fail to detect DTMF over direct ISDN pri link
Title: Fail to detect DTMF over direct ISDN pri link Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the monitor command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. Sylvain Gagnon Speech Technology Integrator BCE Elix Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri lin k
Title: RE: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link Thank you Peter for you reply, I realize this problem occur because I take the CVS head (maybe a bugs get introduce), because when I rebuild using the checkout of the latest stable version (cvs checkout -r v1-0), I don't have the problem of detection of DTMF over a direct ISDN pri link. Hopefully this problem with be fixed before the next release of asterisk. Sylvain. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Peter Svensson Sent: Tuesday, February 15, 2005 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fail to detect DTMF over direct ISDN pri link On Tue, 15 Feb 2005, Sylvain Gagnon wrote: I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the monitor command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. I'm not quite sure I understand your setup when the dtmf detection works / does not work. Can you explain what is connected to what for the two cases? One possibility I can think of is that inband digits are not detected on a PRI until the call is in the PROCEEDING phase. Until then the digits are dent as INFORMATION elements. Perhaps your two pieces of equipment do not agreee on the state of the call? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail to detect DTMF over direct ISDN pri link
Title: Fail to detect DTMF over direct ISDN pri link Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test, I want to used a direct ISDN link with the platform, without to change anything at the Asterix side, I configure the platform to be the ISDN Network side (DMS100) with a twist cable. The D-Channel can up; I am able to perform call, except Asterisk doesn't detect any DTMF anymore? Why? What is the relation with the ISDN link? I use the "monitor" command to record the call, and I really hear the DTMF tone correctly... I try to put relaxdtmf=yes in the Zapata.conf, but no success Thanks for any help or suggestion to diagnose this problem. Sylvain Gagnon, B.Ing., M.Sc.A. Speech Technology Integrator Intégrateur en technologie de la voix BCE Elix Specialist in contact center solutions Spécialiste en solutions pour centres de contacts 14 Commerce Place, 5th Floor Nuns' Island (Québec) CANADA H3E 1T5 t: 514-768-1000, ext. 2224 f: 514-768-7680 www.bceelix.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P problems with 1 FXS, 1 FXO
I am getting a similar problem with a TDM400P with 4 FXO ports, Module 1 not working properly, IE not picking up the calls at all. I have 3 TDM400P cards, 1 loaded with 4 FXO ports, 2 loaded with 4 FXS ports. Does anyone know if this problem with port 1 is only with the FXO modules or if it's also with FXS modules... I'm wondering if I should be moving ports around on the card not to use FXO ports on any port#1, even though it would be pretty confusing when connecting lines on the cards. Also could this problem with port #1 be associated with the Power Alarm on module 1, resetting? Thank you! Sylvain Gagnon [EMAIL PROTECTED] There is a known problem with the first port on the TDM400P. For me, the problem resulted in randomly dropped calls on the FXO port, which was plugged into position 1. Since you only have two daughterboards, try using positions 2-3-4 and don't use position 1, and see of that solves your problem. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Wed, 19 May 2004, David Creemer wrote: Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel modprobe wcfxs report no errors. box:/etc/asterisk# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. --- So it looks like things are OK so far. Here's the relevant portion of my zaptel.conf: defaultzone=us # load FXO X100P channel 1, kewlstart signalling # turned off, card removed #fxsks=1 # load FXS TDM400P channel 1, kewlstart signalling fxoks=1 # load FXO TDM400P channel 2, kewlstart signalling fxsks=2 And here's what dmesg reports: Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 12 for device 00:09.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Not installed Module 2: Installed -- AUTO FXO Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) --- the relevant portions of my zapata.conf are: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes busydetect=yes callprogress=yes ; interfaces for internal analog phones signalling=fxo_ks threewaycalling=yes transfer=yes group=1 context=from-internal callerid=Creemer 01 channel = 1 mailbox=01 ; interfaces to the external PSTN line signalling=fxs_ks context=from-pstn group=2 channel = 2 --- starting asterisk gives: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found May 19 10:42:20 DEBUG[1024]: chan_zap.c:1077 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXO Kewlstart signalling May 19 10:42:20 WARNING[1024]: chan_zap.c:665 zt_open: Unable to specify channel 2: No such device May 19 10:42:20 ERROR[1024]: chan_zap.c:5340 mkintf: Unable to open channel 2: No such device here = 0, tmp-channel = 2, channel = 2 May 19 10:42:20 ERROR[1024]: chan_zap.c:7376 setup_zap: Unable to register channel '2' May 19 10:42:20 WARNING[1024]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 -- Unregistered channel 2 May 19 10:42:20 WARNING[1024]: loader.c:408 load_modules: Loading module chan_zap.so failed! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
