http://blog.gmane.org/gmane.comp.telephony.pbx.asterisk.user hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220058 _______________________________________________ Giedrius Augys 2008-12-01T12:15:15 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220045 _______________________________________________ Olivier 2008-12-01T09:09:06 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220043 _______________________________________________ krunal soni 2008-12-01T08:29:12 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220042 _______________________________________________ Olivier 2008-12-01T08:12:31 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220041 _______________________________________________ Olivier 2008-12-01T07:39:53 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220033 Yesterday I pulled in the latest svn of Dahdi and added the files from a recent kernel in the drivers/staging/echo structure and modified the Kbuild file so it would compile without error. I insmod'ed the module in, and modified my system.conf has echocanceller=oslec. cat /proc/dahdi/1 shows: Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) IRQ misses: 1 1 WCTDM/0/0 FXSKS (In use) (EC: OSLEC) 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 With the reco's from http://www.rowetel.com/ucasterisk/oslec.html#install on configuring the chan_dahdi.conf file, the system behaves exactly as if there is no ec enabled at all? Are there any additional steps needed to enable oslec under dahdi, I am guessing I have missed something? Thanks, jlc _______________________________________________ Joseph L. Casale 2008-11-30T23:02:17 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220021 Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb _______________________________________________ Robert Boardman 2008-11-30T17:27:00 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220014 I have an issue with Dahdi trunk and Asterisk 1.6.0.1 where my analog line is call forwarded on no answer or busy to my sip provider. When we call in on the analog line, I can see the call begin in the cli, and after 15 seconds I see the call switch over to my sip provider, and after about 30 seconds I get the 3 raising tone signals and the call is hungup. Is that my telco dropping the call for some reason? Incoming calls from the sip provider continue on through its context fine if the call originates through it? I assume the transfer to my sip provider happens as my telco decides it needs to do this. I can investigate that Monday, but why doesn't the incoming sip call continue on through the incoming sip dialplan like it does if I call that number directly and get to voicemail after 45 seconds? Is it possible to make Asterisk answer the incoming dahdi call so the Telco is satisfied but provide ringing to the incoming caller until a handset internally answers or it hits voicemail? Thanks! jlc _________ Joseph L. Casale 2008-11-30T00:39:18 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220013 ---cut------------------------------------------------- -= Info about function 'PP_EACH_USER' =- [Syntax] PP_EACH_USER(<string>|<exclude_mac>) [Synopsis] Generate a string for each phoneprov user [Description] Pass in a string, with phoneprov variables you want substituted in the format of %%%%{VARNAME}, and you will get the string rendered for each user in phoneprov excluding ones with MAC address <exclude_mac>. Probably not useful outside of res_phoneprov. Example: ${PP_EACH_USER(<item><fn>%%%%{DISPLAY_NAME}</fn></item>|${MAC}) ---cut------------------------------------------------- The description is a bit vague. "Generate" = return? "for each" = concatenate? ---cut------------------------------------------------- -= Info about function 'PP_EACH_EXTENSION' =- [Syntax] PP_EACH_EXTENSION(<mac>|<template>) [Synopsis] Execute specified template for each extension [Description] Output the specified template for each extension associated with the specified MAC address. ---cut----------------------- Philipp Kempgen 2008-11-29T22:54:23 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/220001 I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a quotation. Thanks Julian. ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________ _______________________________________________ Julian Lyndon-Smith 2008-11-29T13:19:19 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219997 libspandsp.so.0: cannot open shared object file: No such file or directory Created the symlink: /usr/local/lib# ls -lt lib* lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 -> libspandsp.so.1.0.0 -rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a -rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so -> libspandsp.so.1.0.0 lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so.1 -> libspandsp.so.1.0.0 -rwxr-xr-x 1 root staff 1433877 2008-11-13 13:26 libspandsp.so.1.0.0 Edited /etc/ld.so.conf: # Begin ------ /etc/ld.so.conf include /etc/ld.so.conf.d/*.conf /usr/local/lib # End: ------- /etc/ld.so.conf Googled the heck out of it: <http://www.google.com/search?q=libspandsp.so.0:+cannot+open+shared+object> Still can't find the answer. Any ideas? _______________________________________________ Doug 2008-11-29T05:43:22 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219996 _______________________________________________ Yuri 2008-11-29T04:21:16 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219995 _______________________________________________ Yuri 2008-11-29T04:18:50 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219994 _______________________________________________ Sun xiaoshuang 2008-11-29T03:02:43 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219980 _______________________________________________ Robert Augustyn 2008-11-28T19:06:43 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219973 _______________________________________________ Cesc Santa 2008-11-28T16:20:49 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219969 _______________________________________________ Mike 2008-11-28T16:00:40 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219968 I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to PSTN. I initially thought it may be a NAT problem and have checked everything on the NAT gateway/firewall. I see no rejected packets hitting the firewall logs. I'm really at a loss as to what could be causing the calls to drop out for one party so regularly. Any clues where I could look further to debug this would be most useful. local firewall: modprobe ip_conntrack_sip ports=5060 modprobe ip_nat_sip # probably not needed since everything is forwarded: $IPTABLES -A FORWARD -s $INTERNAL_NET -d $ANYWHERE -p udp --dport 5060 -j accept-log # sip remote Asterisk server: $MODPROBE ip_conntrack $MODPROBE ip_conntrack_ Simon Tennant 2008-11-28T15:56:02 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219967 _______________________________________________ michel freiha 2008-11-28T15:07:47 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219959 I had two definitions in sip.conf like: [geishp64_to_mpgeisjhome1] username=geishp64_to_mpgeisjhome1 secret=26 characters long host=x.y.z.1 [geishp64_to_mpgeisjhome2] username=geishp64_to_mpgeisjhome2 secret=26 characters long host=x.y.z.1 When I had BOTH of the above defines in sip.conf seems like 1.4.21 was getting confused on which one to use. auth didnt match was there error I saw. However, if I remove the second entry everything works as expected. (but I do want two, or more entries for testing). Is there a name length limit on username, secret or [X]? Is it an issue that both connections are to the same IP address or host or are my names to long? I thought everything was 256? Thanks, Jerry _______________________________________________ Jerry Geis 2008-11-28T13:32:24 http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/219956 _______________________________________________ equis software 2008-11-28T11:13:31 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.user