[Asterisk-Users] Setting up asterisk on Rh 9
Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _ Expand your wine savvy and get some great new recipes at MSN Wine. http://wine.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up asterisk on Rh 9
FRANCISCO PEREZ-LANDAETA wrote: Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, My install guide may help.. http://members.lycos.co.uk/wipe_out/asterisk Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT How to do it.
Hi I have SIP working on NAT using X-lite phones. On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my * (10.1.0.0). 16394,16384 being RTP. In X-lite set the RTP port to use 16394 instead of the default 8000. Works great over the internet. Didn't need patches or anything else. I hope that helps you. -C www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 27 December 2003 08:34 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi All, i tried to apply this patch and i got the following error. The chan_sip.c version i hv is 1.265 hv any one tried this patch on this latest chan_sip version. thanks, -B chan_sip.o: In function `load_module': chan_sip.o(.text+0x15ebf): undefined reference to `ast_rtp_proto_register' chan_sip.o(.text+0x15ee0): undefined reference to `ast_register_application' chan_sip.o: In function `delete_users': chan_sip.o(.text+0x15fc1): undefined reference to `ast_free_ha' chan_sip.o(.text+0x1604d): undefined reference to `ast_sched_del' chan_sip.o: In function `prune_peers': chan_sip.o(.text+0x16167): undefined reference to `ast_sched_del' chan_sip.o(.text+0x1618d): undefined reference to `ast_sched_del' chan_sip.o: In function `unload_module': chan_sip.o(.text+0x162bd): undefined reference to `ast_channel_unregister' chan_sip.o(.text+0x162ce): undefined reference to `ast_unregister_application' chan_sip.o(.text+0x16337): undefined reference to `ast_softhangup' chan_sip.o(.text+0x1636c): undefined reference to `ast_log' chan_sip.o(.text+0x163ab): undefined reference to `pthread_cancel' chan_sip.o(.text+0x163be): undefined reference to `pthread_kill' chan_sip.o(.text+0x163d1): undefined reference to `pthread_join' chan_sip.o(.text+0x16418): undefined reference to `ast_log' chan_sip.o(.text+0x164b8): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [chan_sip.so] Error 1 - Original Message - From: listas iPfone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 2:10 AM Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi The version 1.260 of chan_sip.c already have that patch?: http://bugs.digium.com/file_download.php?file_id=430type=bug thanks! Miklos - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 2:10 AM Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you
Re: [Asterisk-Users] Interesting problem
On Thu, 18 Dec 2003, Christopher J. Wolff wrote: I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. If you call the Dial application without the r option, that should be fixed. The Skinny driver has a funny way of providing Ringback: It just switches audio through while ringing... If you still need ringback, try doing it explicitly like this: exten = 123,1,Ringing exten = 123,2,Dial(Skinny/[EMAIL PROTECTED]) ; do NOT add |r or ,r here!^ HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7912 speed dials
On Sun, 21 Dec 2003, Ludovic Drolez wrote: We have Cisco 7912 phones, and the doc says that I can create up to four speed dial buttons on my phone using the Cisco CallManager. Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? Yes, it's the Skinny Client Control Protocol, aka Skinny or SCCP. There are 2 Asterisk channel drivers supporting this protocol: - chan_skinny, which comes with asterisk This one is quite stable but supportsy nearly nothing except placing or receiving single calls. - chan_sccp, from http://theo.me.uk/pages.shtml?page=sccp This one may crash asterisk if some very odd skinny phone tries to register. (The 7912 may qualify as odd, as it's not officially supported right now.) However, it does support some nice features (soft keys, multiple calls don't crash *, ...) Some features like intercom (paging) or speed dials don't seem to be completely implemented, yet. But the author seems quite responsive to me and the code is easy to read/update. It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D Sure, but some development work still has to be done, first. Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up asterisk on Rh 9
in addition to wipeout's guide, i also found these helpful: http://www.oneunified.net/support/asterisk/ http://www.automated.it/guidetoasterisk.htm .t Message: 9 From: FRANCISCO PEREZ-LANDAETA [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sat, 27 Dec 2003 09:20:03 + Subject: [Asterisk-Users] Setting up asterisk on Rh 9 Reply-To: [EMAIL PROTECTED] Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _ Expand your wine savvy and get some great new recipes at MSN Wine. http://wine.msn.com --__--__-- Message: 10 Date: Sat, 27 Dec 2003 09:39:33 + From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Setting up asterisk on Rh 9 Reply-To: [EMAIL PROTECTED] FRANCISCO PEREZ-LANDAETA wrote: Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, My install guide may help.. http://members.lycos.co.uk/wipe_out/asterisk Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vocera Communication Badge
Hi there, yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp ** Wireless Specifications Network Standard IEEE 802.11b Frequency Band 2400-2483.4 MHz Data Rates Supported 1, 2, 5.5 and 11 Mbps Wireless Medium Direct Sequence Spread Spectrum (DSSS) Media Access Protocol Carrier sense multiple access with collision avoidance (CSMA/CA) Modulation DBPSK@ 1 Mbps DQPSK@ 2 Mbps CCK@ 5.5 and 11 Mbps Operating Channels 11 channels, 3 non-overlapping Roaming IEEE 802.11b compliant WLAN Security Encryption: 64, 128 WEP, Cisco TKIP Authentication: Open, Cisco LEAP ** Badge Specifications Physical Dimensions 4.2 x 1.4 x .6 in. (10.6 x 3.5 x 1.5 cm) Weight 1.9 oz. (53.9 g), with standard battery pack LED Indicators Two Indicators: single and two-color LCD Supports 4 lines of text, 14 characters per line Controls Call button Hold/Do Not Disturb button Volume/Menu Selection buttons Headset Support 2.5 mm gold plated jack Compatible with Plantronics M175 and M205 headsets Compatible with UmeVoice theBoom noise-canceling headset Electrical Speaker Horn (available December 2003) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
For those who dont speak German: http://tinyurl.com/2ltft Philipp von Klitzing wrote: Hi there, yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp ** Wireless Specifications Network Standard IEEE 802.11b Frequency Band 2400-2483.4 MHz Data Rates Supported 1, 2, 5.5 and 11 Mbps Wireless Medium Direct Sequence Spread Spectrum (DSSS) Media Access Protocol Carrier sense multiple access with collision avoidance (CSMA/CA) Modulation DBPSK@ 1 Mbps DQPSK@ 2 Mbps CCK@ 5.5 and 11 Mbps Operating Channels 11 channels, 3 non-overlapping Roaming IEEE 802.11b compliant WLAN Security Encryption: 64, 128 WEP, Cisco TKIP Authentication: Open, Cisco LEAP ** Badge Specifications Physical Dimensions 4.2 x 1.4 x .6 in. (10.6 x 3.5 x 1.5 cm) Weight 1.9 oz. (53.9 g), with standard battery pack LED Indicators Two Indicators: single and two-color LCD Supports 4 lines of text, 14 characters per line Controls Call button Hold/Do Not Disturb button Volume/Menu Selection buttons Headset Support 2.5 mm gold plated jack Compatible with Plantronics M175 and M205 headsets Compatible with UmeVoice theBoom noise-canceling headset Electrical Speaker Horn (available December 2003) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
ok, fine, if the site goes belly up... http://translate.google.com/translate?u=http%3A%2F%2Fwww.heise.de%2Fnewsticker%2Fdata%2Ftol-25.12.03-001%2Flangpair=de%7Cenhl=enie=UTF8oe=UTF8 Doug Heckaman III wrote: For those who dont speak German: http://tinyurl.com/2ltft Philipp von Klitzing wrote: Hi there, yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp ** Wireless Specifications Network Standard IEEE 802.11b Frequency Band 2400-2483.4 MHz Data Rates Supported 1, 2, 5.5 and 11 Mbps Wireless Medium Direct Sequence Spread Spectrum (DSSS) Media Access Protocol Carrier sense multiple access with collision avoidance (CSMA/CA) Modulation DBPSK@ 1 Mbps DQPSK@ 2 Mbps CCK@ 5.5 and 11 Mbps Operating Channels 11 channels, 3 non-overlapping Roaming IEEE 802.11b compliant WLAN Security Encryption: 64, 128 WEP, Cisco TKIP Authentication: Open, Cisco LEAP ** Badge Specifications Physical Dimensions 4.2 x 1.4 x .6 in. (10.6 x 3.5 x 1.5 cm) Weight 1.9 oz. (53.9 g), with standard battery pack LED Indicators Two Indicators: single and two-color LCD Supports 4 lines of text, 14 characters per line Controls Call button Hold/Do Not Disturb button Volume/Menu Selection buttons Headset Support 2.5 mm gold plated jack Compatible with Plantronics M175 and M205 headsets Compatible with UmeVoice theBoom noise-canceling headset Electrical Speaker Horn (available December 2003) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up asterisk on Rh 9
thanks.. i will try it out.. I really appreciated... Francisco From: WipeOut [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Setting up asterisk on Rh 9 Date: Sat, 27 Dec 2003 09:39:33 + FRANCISCO PEREZ-LANDAETA wrote: Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, My install guide may help.. http://members.lycos.co.uk/wipe_out/asterisk Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Working moms: Find helpful tips here on managing kids, home, work and yourself. http://special.msn.com/msnbc/workingmom.armx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
Hi there, yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it in real life), and if a price tag can be found for this device? Too bad they don't use SIP... ;-( http://www.vocera.com/ http://www.heise.de/newsticker/data/tol-25.12.03-001/ Cheers, Philipp Looks interesting. I seem to remember something a while back regarding VoiceXML and the TellMe folks. Maybe there is something that is open sourced regarding speach recognition on the server side. An open sourced hardware, as recently discussed, should be able to be packaged so that it could also be a wearable device even if part is on the belt and the speech I/O as separate. Will check with come contacts and see if anyone was involved in the U.S.S. Coronado tests. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Incoming callers aren't hearing ring
you need to answer the line to place audio on the channel. So if you place an answer line before the dials, you should get audio to route back. I just changed extensions.conf to read: /etc/asterisk/extensions.conf [incoming] include = sip-phones exten = _5551212,1,Answer exten = _5551212,2,Dial(SIP/6710,12,tr) exten = _5551212,3,Dial(SIP/6710SIP/6711SIP/6712SIP/6713,20,tr) exten = _5551212,4,Voicemail2(u6710) exten = _5551212,5,Hangup exten = _5551212,104,Voicemail2(b6710) exten = _5551212,105,Hangup and restarted asterisk for good measure. I am still not getting any ring on the calling end. Anything else I should check? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
thats cool. i ll try that too. Whats ur * version. if thats the case what is this patch for. Is bug 104 already approved and in production. -B - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:43 AM Subject: RE: [Asterisk-Users] Asterisk behind NAT How to do it. Hi I have SIP working on NAT using X-lite phones. On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my * (10.1.0.0). 16394,16384 being RTP. In X-lite set the RTP port to use 16394 instead of the default 8000. Works great over the internet. Didn't need patches or anything else. I hope that helps you. -C www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 27 December 2003 08:34 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi All, i tried to apply this patch and i got the following error. The chan_sip.c version i hv is 1.265 hv any one tried this patch on this latest chan_sip version. thanks, -B chan_sip.o: In function `load_module': chan_sip.o(.text+0x15ebf): undefined reference to `ast_rtp_proto_register' chan_sip.o(.text+0x15ee0): undefined reference to `ast_register_application' chan_sip.o: In function `delete_users': chan_sip.o(.text+0x15fc1): undefined reference to `ast_free_ha' chan_sip.o(.text+0x1604d): undefined reference to `ast_sched_del' chan_sip.o: In function `prune_peers': chan_sip.o(.text+0x16167): undefined reference to `ast_sched_del' chan_sip.o(.text+0x1618d): undefined reference to `ast_sched_del' chan_sip.o: In function `unload_module': chan_sip.o(.text+0x162bd): undefined reference to `ast_channel_unregister' chan_sip.o(.text+0x162ce): undefined reference to `ast_unregister_application' chan_sip.o(.text+0x16337): undefined reference to `ast_softhangup' chan_sip.o(.text+0x1636c): undefined reference to `ast_log' chan_sip.o(.text+0x163ab): undefined reference to `pthread_cancel' chan_sip.o(.text+0x163be): undefined reference to `pthread_kill' chan_sip.o(.text+0x163d1): undefined reference to `pthread_join' chan_sip.o(.text+0x16418): undefined reference to `ast_log' chan_sip.o(.text+0x164b8): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [chan_sip.so] Error 1 - Original Message - From: listas iPfone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 2:10 AM Subject: Re: [Asterisk-Users] Asterisk behind NAT How to do it. Hi The version 1.260 of chan_sip.c already have that patch?: http://bugs.digium.com/file_download.php?file_id=430type=bug thanks! Miklos - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 2:10 AM Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd
[Asterisk-Users] Help with x101P
Hello, I recently purchased an x101p and am having problems with it. I get random hangups after which the channel provisioned for this card is unsuable (until the driver is uninstalled and then reinstalled). Looiking in the log file (and running zttool) I can see a red alarm on the card even though their is connectivity to walljack. I have read the archives and have experimented with tweaking busydetect and callprogress params with no luck. Any help is greatly appreciated. thanks burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
busydetect and callprogress should be turned off in my experience, enabling any of those parameters causes random drops. comment the lines out of your config file, and try again. Mark At 02:32 PM 12/27/2003 -0400, Burak Balasaygun wrote: Hello, I recently purchased an x101p and am having problems with it. I get random hangups after which the channel provisioned for this card is unsuable (until the driver is uninstalled and then reinstalled). Looiking in the log file (and running zttool) I can see a red alarm on the card even though their is connectivity to walljack. I have read the archives and have experimented with tweaking busydetect and callprogress params with no luck. Any help is greatly appreciated. thanks burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call with bad/choppy sound
Hi all. I have this configuration: Telco -(E1)-TE410P//Dual Xeon Server 2.4Ghz-(Ethernet)-Switch-GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS receives calls it sounds great. 2.- We are not receiving any callerd id from the PSTN, this may be an issue with the E1 provider, will checkit, but again, it might not. Thank's in advance for any help. Configuration: /etc/zaptel.conf -- span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone=us defaultzone=us /etc/asterisk/zapata.conf -- [channels] callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no amaflags=default switchtype=5ess signalling=pri_cpe pridialplan=unknown context=default usecallerid=yes hidecallerid=no callerid=asreceived channel=1-15,17-31 /etc/asterisk/sip.conf - [general] port=5060 bindaddr=0.0.0.0 externip='some-ip' context=default disallow=all allow=gsm allow=ulaw allow=alaw allow=g729 tos=lowdelay register=67373:'passwd'@fwd.pulver.com/0100 [fwd.pulver.com] type=friend secret='passwd' username=67373 host=fwd.pulver.com context=default nat=yes fromuser=67373 fromdomain=fwd.pulver.com reinvite=no canreinvite=no qualify=500 ;The GS/BT [pedro] type=friend host=dynamic dtmfmode=inband mailbox=320 username=pedro secret='passwd' nat=no callgroup=1 pickupgroup=1 disallow=all allow=ulaw callerid=Pedro 320 [67373] type=friend host=fwd.pulver.com context=default nat=yes fromdomain=fwd.pulver.com reinvite=no canreinvite=no qualify=500 disallow=all allow=all ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with x101P
How do you have your zaptel.conf set up (please post it here). Also, what type of switch are you connected to? Regards, Scott M. Stingel Emerging Voice Technology Inc. Email: scott a t evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Burak Balasaygun Sent: Saturday, December 27, 2003 6:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with x101P Hello, I recently purchased an x101p and am having problems with it. I get random hangups after which the channel provisioned for this card is unsuable (until the driver is uninstalled and then reinstalled). Looiking in the log file (and running zttool) I can see a red alarm on the card even though their is connectivity to walljack. I have read the archives and have experimented with tweaking busydetect and callprogress params with no luck. Any help is greatly appreciated. thanks burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
Hi! Mahoney, Matt wrote: Are there any hardware SIP phones that have IPsec or other encryption built in? I knew of a few vendors that had products under development (as of August). I don't know where things stand now however. Others might. You will probably not see IPsec support however. The VoIP world is backing SRTP (bearer) and TLS (SIP signaling) over IPsec. A lot has to do with IKE or, key management in general. The Zultys 4x4 comes with SRTP and AES encryption. http://www.voip-info.org/wiki-Zultys+phones Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mysql cdrs
How can I download the asterisk-addons and setup CDR support for mysql? I reviewed the wiki but did not find instructions on dowloading. Just a sample of the cdr_mysql.conf file. DaL -- David A. Lauer Network Engineer Tristar Communications [EMAIL PROTECTED] 954.977.8081 ext. 21 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mysql cdrs
You can check it out via CVS. asterisk-addons -Original Message- From: David A. Lauer [mailto:[EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:16 PM To: Asterisk Users Subject: [Asterisk-Users] mysql cdrs How can I download the asterisk-addons and setup CDR support for mysql? I reviewed the wiki but did not find instructions on dowloading. Just a sample of the cdr_mysql.conf file. DaL -- David A. Lauer Network Engineer Tristar Communications [EMAIL PROTECTED] 954.977.8081 ext. 21 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mysql cdrs
Yes, cvs checkout asterisk-addons, then go into the resultant source directory and make clean; make install. You also have to setup the cdr_mysql.conf file, as per the sample provided. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Saturday, December 27, 2003 8:21 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] mysql cdrs You can check it out via CVS. asterisk-addons -Original Message- From: David A. Lauer [mailto:[EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:16 PM To: Asterisk Users Subject: [Asterisk-Users] mysql cdrs How can I download the asterisk-addons and setup CDR support for mysql? I reviewed the wiki but did not find instructions on dowloading. Just a sample of the cdr_mysql.conf file. DaL -- David A. Lauer Network Engineer Tristar Communications [EMAIL PROTECTED] 954.977.8081 ext. 21 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
Hi again! yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it Looks interesting. I seem to remember something a while back regarding VoiceXML and the TellMe folks. Maybe there is something that is open sourced regarding speach recognition on the server side. Now I also found an interesting review comparing client solutions for VoIP throug WLAN (let's add a wireless keyword for the search engines): http://www.kinetowireless.com/news/industry_articles/vowlan_takeoff.html This includes a secion on the Vocera. The concept as such surely deserves some aaahs and hs... :-) Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mysql cdrs
Or use cdr_odbc :P On Sat, 27 Dec 2003, Scott Stingel wrote: Yes, cvs checkout asterisk-addons, then go into the resultant source directory and make clean; make install. You also have to setup the cdr_mysql.conf file, as per the sample provided. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Saturday, December 27, 2003 8:21 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] mysql cdrs You can check it out via CVS. asterisk-addons -Original Message- From: David A. Lauer [mailto:[EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:16 PM To: Asterisk Users Subject: [Asterisk-Users] mysql cdrs How can I download the asterisk-addons and setup CDR support for mysql? I reviewed the wiki but did not find instructions on dowloading. Just a sample of the cdr_mysql.conf file. DaL -- David A. Lauer Network Engineer Tristar Communications [EMAIL PROTECTED] 954.977.8081 ext. 21 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] frame buffering
Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
On Sat, 2003-12-27 at 15:30, Ing. Angel Gomez Garcia wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. Turn it off. In a text only install you only get prettier text. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
On Sat, 2003-12-27 at 14:13, Philipp von Klitzing wrote: Hi! Mahoney, Matt wrote: Are there any hardware SIP phones that have IPsec or other encryption built in? I knew of a few vendors that had products under development (as of August). I don't know where things stand now however. Others might. You will probably not see IPsec support however. The VoIP world is backing SRTP (bearer) and TLS (SIP signaling) over IPsec. A lot has to do with IKE or, key management in general. The Zultys 4x4 comes with SRTP and AES encryption. http://www.voip-info.org/wiki-Zultys+phones I've been meaning to ask, is the aes stuff recently checked in part going to implement srtp or iax with aes? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
- Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:58 PM Subject: Re: [Asterisk-Users] Vocera Communication Badge Hi again! yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it Looks interesting. I seem to remember something a while back regarding VoiceXML and the TellMe folks. Maybe there is something that is open sourced regarding speach recognition on the server side. Now I also found an interesting review comparing client solutions for VoIP throug WLAN (let's add a wireless keyword for the search engines): http://www.kinetowireless.com/news/industry_articles/vowlan_takeoff.html This includes a secion on the Vocera. The concept as such surely deserves some aaahs and hs... :-) Ooohs and Aaahs are better than the drool my wife wiped away from my mouth after I showed her the link :-) Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
James Sharp wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Thanks. How do I verify that my console is running frame buffering ? and mos important, How do I disable it ? What should I do to run a plain VGA console ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vocera Communication Badge
- Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 5:53 PM Subject: Re: [Asterisk-Users] Vocera Communication Badge - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 27, 2003 3:58 PM Subject: Re: [Asterisk-Users] Vocera Communication Badge Hi again! yesterday I came across the Vocera Communication Badge and now I'd like to know if anyone here has played with that thing (or even just seen it Looks interesting. I seem to remember something a while back regarding VoiceXML and the TellMe folks. Maybe there is something that is open sourced regarding speach recognition on the server side. Now I also found an interesting review comparing client solutions for VoIP throug WLAN (let's add a wireless keyword for the search engines): http://www.kinetowireless.com/news/industry_articles/vowlan_takeoff.html This includes a secion on the Vocera. The concept as such surely deserves some aaahs and hs... :-) Ooohs and Aaahs are better than the drool my wife wiped away from my mouth after I showed her the link :-) Andrew Thompson http://aktzero.com/ OK, for the non off-topic response, In the link above, I found this statement: For those who find standards important to their enterprise, Vocera does not currently use SIP, but plans to add limited support for it in the future. Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
On Sat, 2003-12-27 at 16:28, Ing. Angel Gomez Garcia wrote: James Sharp wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. How do I verify that my console is running frame buffering ? and mos important, How do I disable it ? What should I do to run a plain VGA console ? Is there a penguin in the upper left when it boots, or some other graphic? If so your in a frame buffer. To disable requires recompiling the kernel and removing the option. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help - after last night update unable to make outbound calls from GS
Hi All, i updated * last night. And after that i am unable to make any outbound calls from GS. all my config files are same no change. When ever i try to make a call from GS i constantly get this warning in * console. Warning: Filechan_sip.c line471 Maximum retires exceeded on call some guid 192.168.0.22 for seqno 2126 (response). i rebooted my GS, restarted * no change. i did a search for this error message - the response i found was in sip.conf one shd hv disallow=all allow=ulaw allow=alaw allow=libc allow=gsm i do hv these entries in my sip.conf. Incoming calls to this GS work fine. Any idea what could be the problem. thanks, -B Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
Re: [Asterisk-Users] frame buffering
On Saturday 27 December 2003 16:42, Steven Critchfield wrote: On Sat, 2003-12-27 at 16:28, Ing. Angel Gomez Garcia wrote: James Sharp wrote: Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. How do I verify that my console is running frame buffering ? and mos important, How do I disable it ? What should I do to run a plain VGA console ? Is there a penguin in the upper left when it boots, or some other graphic? If so your in a frame buffer. To disable requires recompiling the kernel and removing the option. Actually, it's even easier than that to disable: If you're using LILO: 1) Edit lilo.conf and remove any line that begins with vga= 2) run /sbin/lilo -v 3) reboot -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dual Athlon 2.4 MP *
Take a glance at bugs 714 thru 722 on bugs.digium.com I feel this is a local hardware issue. Has anyone else ran on a dual amd box ? Could his power supply be too weak? I don't know of anyone that has 9 totally diffrent and totally random crashes in asterisk in one day. Anyone care to input on this? Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql cdrs
cdr_odbc is for logging CDR data to a database. Its pretty much blind to the type of database you choose as long as it has an ODBC driver. We had it speaking to an AS/400 running DB2... we also have it working with MSSQL (not my goal but hey it works), mysql, pgsql and flatfiles. I have yet to hear it works with oracle (anyone out there test this?) bkw On Sat, 27 Dec 2003, David A. Lauer wrote: Thanks, I might be interested. If I use cdr_odbc what do I gain or loose? Any codec limitations? Performance or reporting limitations? Message: 10 Date: Sat, 27 Dec 2003 15:10:14 -0600 (CST) From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] mysql cdrs Reply-To: [EMAIL PROTECTED] Or use cdr_odbc :P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with x101P
Scott, Posted below are both zaptel.conf and zapata.conf as they were configured when the problem occured.I have since changed the busydetect parameter to have a value of no in light of Mark's advice. The problem has not recurred (but it has been a very short while since I made the change) since making the change. I'm not sure what you mean by what type of switch you are connected to? The x101p is connected to the CO switch for my LEC. many thanks burak zaptel.conf fxsks=1 fxoks=2 loadzone=us defaultzone=us zapata.conf -- ; ; Zapata telephony interface ; ; Configuration file [channels] busydetect=no busycount=5 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 context=home group=2 signalling=fxo_ks mailbox=0273 callerid=Phone 1 0273 channel=2 On Sat, 27 Dec 2003 19:57:36 -, Scott Stingel wrote How do you have your zaptel.conf set up (please post it here). Also, what type of switch are you connected to? Regards, Scott M. Stingel Emerging Voice Technology Inc. Email: scott a t evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Burak Balasaygun Sent: Saturday, December 27, 2003 6:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with x101P Hello, I recently purchased an x101p and am having problems with it. I get random hangups after which the channel provisioned for this card is unsuable (until the driver is uninstalled and then reinstalled). Looiking in the log file (and running zttool) I can see a red alarm on the card even though their is connectivity to walljack. I have read the archives and have experimented with tweaking busydetect and callprogress params with no luck. Any help is greatly appreciated. thanks burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: time to build an open phone?
While looking for some cell phone goodies to go with my ngage, I noticed a bluetooth earpiece that happened to come with a usb adapter. It made me think of this project. Many of the bluetooth adapters have a single button used on cell phones for voice dialing and answer/hangup functions. Doesn't this sound a lot like a good tie in with a gnophone or other iax client. It also sounds like a good idea as the hardware is already around and functional. While bluetooth is already being supported in linux, I'm not impressed yet with the glue software. I bought a bluetooth adapter so I could try and sync wirelessly the contacts on my phone. Also even with the crap you have to go through it is the best way to transfer apps to my phone. Well that should plug the hardware hole for temporary, go dig into the software. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple mpg123 processes when starting asterisk
When I start asterisk, it appears that multiple mpg123 processes start. Would this be normal operation? 2729 ? S 0:00 /usr/sbin/asterisk 2735 ? S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2736 ? S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2740 ? S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 2742 ? S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 2755 pts/1 R 0:00 ps -ax
Re: [Asterisk-Users] Multiple mpg123 processes when starting asterisk
two for each music on hold class. bkw On Sun, 28 Dec 2003, Kevin wrote: When I start asterisk, it appears that multiple mpg123 processes start. Would this be normal operation? 2729 ?S 0:00 /usr/sbin/asterisk 2735 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2736 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 av-1.mp3 2740 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 2742 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -z av-1.mp3 2755 pts/1R 0:00 ps -ax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
