Re: [Asterisk-Users] Weird issues with Asterisk@home 0.9
See inline reply: On Sat, 16 Apr 2005, Henry Devito wrote: > > 1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro) > > softphone. I can receive voicemail no problem and even in this revision > > the MWI seems to work correctly, though when i try to go to the message > > center, (*98) and enter my voicemail box number, it always gives me > > invalid password. > > If i now do this from a analog phone which is also connected to the > > system via a TDM400P card, i can get into the voicemail box no problem. > > Anyone have experienced this before? > > > > What do you have the DTMF type set for in your SIP.conf and on the phone? > It should be RTP (RFC2833) The password problem seems to be fixed on the voicemail, though dtmf settings didn't change anything and was set by default on both phone and asterisk server. What i had to do to get the sip channel/voicemail to work was delete the setting and add it again. It seemed to have fixed this problem. > > 2) Another problem i came accross is when using the IAX2 extentions (using > > IAX phone), and trying to make a call i get nothing but static on the > > line, even between IAX <-> IAX and IAX <-> SIP, IAX -> PSTN.. This wasn't > > an issue in the 0.8 release where it seemed quite stable and the sound > > quality was even better then from the SIP phone. Got rid of some static, by accident, thus don't know exactly what i did there, but it still isn't as clean as it was with 0.8 > > > > 3) The next problem i have encountered is that the webmail feature doesn't > > seem to work right, as in accepting user input to log in. It seems as if > > the mysql tables/inputs haven't been setup correctly. I have tried the > > webmail with multiple different userid's, and always get username/password > > invalid. Any ideas? > > > > > > 4) The last problem, though might not be specific to AAH, i am still > > experiencing a sound issue when calling the PSTN, where the > > connection sounds like the rushing of an ocean in the phone line, and > > a light echo every once in a while. I live in Canada, and use Telus as my > > Telco provider, so i am not sure what specifics i may be missing. > > I have totally rewired the house for asterisk, thus new lines everywhere, > > making sure they are not to close to powerlines, etc for static concerns. > > Also standard phones don't seem to have the problem. > > > > I should mention, this problem occurs both going through SIP (eyeBeam) > > aswell as a Analog phone (Siemens Gigaset SL30 connected via FXS port). I have tried now messing with the gain settings and it seems to be having no effect what soever. I checked that the TDM22B card has its own IRQ, (checking both lspci -v and /proc/interrupts). Did a zttest and got the following results: Best: 100.00 -- Worst: 99.987793 , which seem pretty good. Still i hear a light echo and some massive ocean like static in the line. Line also sounds sort of muffeled.. Sascha ___ 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] Weird issues with Asterisk@home 0.9
Hi, I am having a few issue withs [EMAIL PROTECTED] 0.9. 1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro) softphone. I can receive voicemail no problem and even in this revision the MWI seems to work correctly, though when i try to go to the message center, (*98) and enter my voicemail box number, it always gives me invalid password. If i now do this from a analog phone which is also connected to the system via a TDM400P card, i can get into the voicemail box no problem. Anyone have experienced this before? What do you have the DTMF type set for in your SIP.conf and on the phone? It should be RTP (RFC2833) 2) Another problem i came accross is when using the IAX2 extentions (using IAX phone), and trying to make a call i get nothing but static on the line, even between IAX <-> IAX and IAX <-> SIP, IAX -> PSTN.. This wasn't an issue in the 0.8 release where it seemed quite stable and the sound quality was even better then from the SIP phone. 3) The next problem i have encountered is that the webmail feature doesn't seem to work right, as in accepting user input to log in. It seems as if the mysql tables/inputs haven't been setup correctly. I have tried the webmail with multiple different userid's, and always get username/password invalid. Any ideas? 4) The last problem, though might not be specific to AAH, i am still experiencing a sound issue when calling the PSTN, where the connection sounds like the rushing of an ocean in the phone line, and a light echo every once in a while. I live in Canada, and use Telus as my Telco provider, so i am not sure what specifics i may be missing. I have totally rewired the house for asterisk, thus new lines everywhere, making sure they are not to close to powerlines, etc for static concerns. Also standard phones don't seem to have the problem. I should mention, this problem occurs both going through SIP (eyeBeam) aswell as a Analog phone (Siemens Gigaset SL30 connected via FXS port). Here is my zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes hanguponpolarityswitch=yes ; Added to test hangups callreturn=yes echocancel=yes echocancelwhenbridged=yes ; was no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no channel=3 ; context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=0 channel=4 ; context=from-internal signalling=fxo_ks usecallerid=asreceived echocancel=yes echocancelwhenbridged=yes echotraining=800 callerid=2892091 group=1 channel=1-2 ;#include zapata-channels.conf Anyone have any ideas? Please let me know. Thanks S. ___ 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] Weird issues with Asterisk@home 0.9
Hi, I am having a few issue withs [EMAIL PROTECTED] 0.9. 1) Setting up a sip connection, with voicemail to use with (eyebeam/X-pro) softphone. I can receive voicemail no problem and even in this revision the MWI seems to work correctly, though when i try to go to the message center, (*98) and enter my voicemail box number, it always gives me invalid password. If i now do this from a analog phone which is also connected to the system via a TDM400P card, i can get into the voicemail box no problem. Anyone have experienced this before? 2) Another problem i came accross is when using the IAX2 extentions (using IAX phone), and trying to make a call i get nothing but static on the line, even between IAX <-> IAX and IAX <-> SIP, IAX -> PSTN.. This wasn't an issue in the 0.8 release where it seemed quite stable and the sound quality was even better then from the SIP phone. 3) The next problem i have encountered is that the webmail feature doesn't seem to work right, as in accepting user input to log in. It seems as if the mysql tables/inputs haven't been setup correctly. I have tried the webmail with multiple different userid's, and always get username/password invalid. Any ideas? 4) The last problem, though might not be specific to AAH, i am still experiencing a sound issue when calling the PSTN, where the connection sounds like the rushing of an ocean in the phone line, and a light echo every once in a while. I live in Canada, and use Telus as my Telco provider, so i am not sure what specifics i may be missing. I have totally rewired the house for asterisk, thus new lines everywhere, making sure they are not to close to powerlines, etc for static concerns. Also standard phones don't seem to have the problem. I should mention, this problem occurs both going through SIP (eyeBeam) aswell as a Analog phone (Siemens Gigaset SL30 connected via FXS port). Here is my zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes hanguponpolarityswitch=yes ; Added to test hangups callreturn=yes echocancel=yes echocancelwhenbridged=yes ; was no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no channel=3 ; context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=0 channel=4 ; context=from-internal signalling=fxo_ks usecallerid=asreceived echocancel=yes echocancelwhenbridged=yes echotraining=800 callerid=2892091 group=1 channel=1-2 ;#include zapata-channels.conf Anyone have any ideas? Please let me know. Thanks S. ___ 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