[Asterisk-Users] dial (iax/Xsip/y) get y fraction earlier
Hello I like to call to 2 providers provider X = IAX provider Y = SIP exten = _06.,1,Dial(IAX2/X/${EXTEN},30,r)(SIP/[EMAIL PROTECTED]) exten = _06.,2,Hangup Provider X is working but provider Y never shows up. What's wrong ?? How can I get provider Y working a fraction earlier the provider X Thanks Sjaak ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] Asterisk RSS list feeder ready
Hello Just for fun a rss newsreader for the asterisk users and biz list. Easy to use and now with the complete history to search. Just use it if you like Thanks Sjaak ___ 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] Asterisk rss list feeder ready
Sorry URL is needed http://asterisk.voipexco.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
[Asterisk-Users] AreskiCC + MySQL
Hello * Users Did somebody get managed to get AreskiCC work under mysql. If so is there anywhere to find the database structure for mysql. Thanks Sjaak ___ 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] RSS feed Asterisk-Users
hello Asterisk-Users I created just for fun a rss feed for Asterisk-Users and Asterisk-Biz list First for myself but if it is usefull for you can use it if you like. But some questions Is it allowed ? If you need add-on's please let me know. When many people will use it I need to generate a little money to pay traffic is it allowed to ad googleadd's ? I've added a search box and later on I will add the whole list history so it will be usefull to search. Just look at http://asterisk.voipexco.com Thanks, and have fun Sjaak ___ 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] bad sound ISDN bristuff
Hello * users I've problems with sound quality on zaphfc Asterisk works fine good sound quality. If I do make load in the bristuf.xx zaphfc dir then sound quality drops directly. Even if I don't load the chan_zap in the modules.conf I use this config on more (even old 400Mhz machines) and works correctly. Looks like an hardware problem but I can't find it. I don't see any conflics on IRQ or interupts Using : Asterisk 1.0.1-BRIstuffed-0.2.0-RC1 ISDN HFC cologne cards If you have any solutions I would like to hear it. cat /proc/interupts --- CPU0 0: 32065843 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 6: 0 XT-PIC ehci_hcd 8: 1 XT-PIC rtc 10: 757932899 XT-PIC zaphfc 12: 14254461 XT-PIC eth0 14:1259128 XT-PIC ide0 NMI: 0 ERR: 0 cat /proc/pci - PCI devices found: Bus 0, device 0, function 0: Host bridge: Silicon Integrated Systems [SiS] SiS651 Host (rev 2). Master Capable. Latency=32. Non-prefetchable 32 bit memory at 0xe800 [0xebff]. Bus 0, device 1, function 0: PCI bridge: Silicon Integrated Systems [SiS] SiS 530 Virtual PCI-to-PCI bridge (AGP) (rev 0). Master Capable. No bursts. Min Gnt=8. Bus 0, device 2, function 0: ISA bridge: Silicon Integrated Systems [SiS] SiS962 [MuTIOL Media IO] (rev 37). Bus 0, device 2, function 5: IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev 0). IRQ 11. Master Capable. Latency=128. I/O at 0xa400 [0xa40f]. Bus 0, device 3, function 0: USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 15). IRQ 3. Master Capable. Latency=32. Max Lat=80. Non-prefetchable 32 bit memory at 0xe700 [0xe7000fff]. Bus 0, device 3, function 1: USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (#2) (rev 15). IRQ 5. Master Capable. Latency=32. Max Lat=80. Non-prefetchable 32 bit memory at 0xe680 [0xe6800fff]. Bus 0, device 3, function 3: USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller (rev 0). IRQ 6. Master Capable. Latency=32. Max Lat=80. Non-prefetchable 32 bit memory at 0xe600 [0xe6000fff]. Bus 0, device 4, function 0: Ethernet controller: Silicon Integrated Systems [SiS] SiS900 10/100 Ethernet (rev 145). IRQ 12. Master Capable. Latency=32. Min Gnt=52.Max Lat=11. I/O at 0x8800 [0x88ff]. Non-prefetchable 32 bit memory at 0xe580 [0xe5800fff]. Bus 0, device 14, function 0: Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 2). IRQ 10. Master Capable. Latency=16. Max Lat=16. I/O at 0x8400 [0x8407]. Non-prefetchable 32 bit memory at 0xe500 [0xe5ff]. Bus 1, device 0, function 0: VGA compatible controller: Silicon Integrated Systems [SiS] SiS65x/M650/740 PCI/AGP VGA Display Adapter (rev 0). IRQ 11. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. Non-prefetchable 32 bit memory at 0xe780 [0xe781]. I/O at 0xd800 [0xd87f]. Thanks anyway Sjaak -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ 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] FAX detection in extentions.conf
Hello I have a PBX for our company with in front a ISDN * ISDN server. I would like to detect if the outgoing line is a fax call or voice call. I know ther's a fax detection in zaptel. Now the question can I make desision in extentions.conf when * detects a fax call. FAX PBX ZAP/G1 ZAP/G2the world and not FAX PBX ZAP/G1 IAX2 voip world Thanks ___ 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] .call MeetMe
Hello Would it be possible to dail out to lett's say to 4 people with a .call file and put them directly into a free meetme room. Thanks Sjaak ___ 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] Help me ($$$) with install h323
Hello Does anybody who have experience in installing the h323 modules in asterisk. I try'd it many times an spend xxx hours to install it but didn't get lucky so far. I have asterisk 1.0.1 running with bristuffed 2.0 I'm willing to pay for this. Sjaak ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN on com port /dev/ttyS0 possible ??
Hello I buyed a new server 2*XEON in a 2inch High 19case. Now I have a problem that the riser card is 64bit so an ISDN PCI modem isn't possible. My question is can I use ISDN on com port /dev/ttyS0. If yes can I use it like the example in modem.conf as /dev/ttyI0 but use /dev/ttyS0 Does anybody have expirience with this ? Thanks Sjaak -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CLI h.323 show codecs shows nothing
Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay I Guess When I make a connection * SIP h323 device, the phone is ringing and then * tells me No one available. and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like. allow=all or disallow=all allow=ulaw allow=gsm Could that be the problem to connecting to an h.323 device ? Can somebody give me a good h323.conf example for outgoing calls to an h323. Or any advice would be nice. I'm getting @#!$$%^#^$%^$$% Cheers Sjaak -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ 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] if NOT SipUser then Dial(Zap/1/${EXTEN})
Hello I like to setup a dial plan where sip users have the same number as there normal phone number. So if sombody dial the number i like that asterisk looks first is it a number who exsist in MySQL sipfriends if not dial out with a ZAP device. I know that extention.conf doesn't have a programming language but something like this if ( SIP/${EXTEN}) { exten = _XX,1,Dial(SIP/${EXTEN}) exten = _XX,2,Hangup } else { exten = _XX,1,Dial(Zap/1/${EXTEN}) exten = _XX,2,Hangup } Does anybody here a solution for this. Thanks. Sjaak ___ 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
Re: [Asterisk-Users] Can some bady help me ???
Hello Are you online with you suse box. If you give me for a ssh login I can look to it. Or otherwise give us more info. Sjaak Rodney Acosta Coya wrote: I get the pakage asterisk-1.0.0.tar.gz but i cant compile a install it For help, i have a suse 9.1 distro in my computer. im new using linux thanks in advance Rodney -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ 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
Re: [Asterisk-Users] grandstream 102 flashing
Check out your DNS settings Wait +60 sec dean collins wrote: Can you (or anyone else out there) tell me how to fix this? Basically what happened was I tried to log into the web interface (ip address allocated by sbs dhcp using mac address) but when I hit login it rebooted and has been doa since then). I checked using MS sbs network monitoring all it seems to be doing is asking for a ARP Rarp request to 67.153.142.69 The other thing is it thinks it is ip address 192.168.1.160 but that isnt even part of my network. Any thoughts on what to do from here? The lcd display is totally non responsive. * From: * [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *BetaTeilchen *Sent:* Wednesday, October 20, 2004 11:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] grandstream 102 flashing This flashing is an indicator for a damaged firmware in your phone. Maybe an interrupted TFTP-Download when powered up or just a wrong firmware. dean collins schrieb: Does anyone know what it means when a grandstream flashes the red key light 5 times repeatedly in cycles? I got a new handset delivered to me today, powered up fine until I tried to access it via the web interface using the password admin and then it rebooted with the lcd never displaying again and the red keys flashing 5 times then a break of 3 seconds then repeat. Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[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
[Asterisk-Users] codec problems with astcc and not with sip trhough aix
Hello . I have a 800 tollfree trhough iax to my * server. If I phone to 800 number to the * machine to a sip phone everything is okay. exten = 8,1,Dial(SIP/12345678,20) -- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format = 4, actual format = 4 If I change my extention to the astcc script i get lott's of these errors exten = 8,1,DeadAGI(astcc.agi) Oct 21 00:58:22 NOTICE[1110648512]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/[EMAIL PROTECTED]:4569/1 of format G729A since our native format has changed to ULAW I see these two difference are codec problems but why. and what to do with this problems When I phone with a second SIP phone to the 8XX number everything is okay to the sipphone and to the astcc script My sip phones accept all codecs My aix.conf looks like this [tollfreeprovider] type=user host=xx.xx.xx.xx secret=mysecret notransfer=yes disallow=all ;allow=g729 ;allow=gsm allow=ulaw trunk=yes context=tollfree I don't know, could this be a problem for my tollfree number provider or is it me. Cheers ___ 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
Re: [Asterisk-Users] ASTCC newbie
I just installed it 2 days ago and I was looking for a long time to get thes problem solved. Be aware off creating the Mysql database the astcc.cgi script will do. But I disable the create lines in the perl script. The astcc.cgi script will endles wait. Good luck ___ 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
Re: [Asterisk-Users] SIP Extensions
Hello Ron Ramos With the power of bash it's easy copy this code to a file name it createsip.sh #!/bin/bash for ((i=2000 ; i 8000; i++ )); do echo [$i] echo secret=$i echo type=friend echo username=$i echo done chmod 700 createsip.sh ./createsip.sh /etc/asterisk/sip.conf and your done Thats all Hi All, How can I be able to define multiple SIP extensions? Do I have to define each extensions on sip.conf? For example, extension 2000-8000, do I have to define it one by one on sip.conf? [2000] secret=2000 type=friend username=user2000 .. .. [2001] secret=2001 type=friend username=user2001 .. .. .. .. .. [8000] Is there a way I can define multiple extensions like... [_2XXX] .. .. .. [_8XXX] ___ 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] sip show peers MySQL Database
Hello How can i see the sip show peers if I use sipfriends database. I see only the peers who are in the sip.conf. Thanks Sjaak ___ 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] SIP providers USA
Hello This Question maybe ask many times. I'm From the Netherlands and looking for an sip/aix pstn provider in the US.I asked wipphone, vonage but they don't work for europe. I've visit many not finished website's about sip pstn Can somebody recomed me any good sip/aix provider in the us. Thanks ___ 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] The best simple solution (newbee)
Hi I'm very new to astrisk and I like to ask you as users with expirience what hardware to use and how to set this up. What I like to create is the followe : Use a normale phone dail a number astrisk (2Mbit) Internet(64Kb) astrisk Normal phone starts to ring and someone pick up the phone and talk and also phone reverse. Ther's so lot of hardware x100p phonejack linejack Cisco and more and more i've not a good idea what's good for me. I've read so much that now my brain is [EMAIL PROTECTED]*()_ My main language isn't English so I understand everything for 90% and that's not enough to buy something like this. I just like to setup a simple and (low budget) configuration. I know that 64K line isn't that much but is it acceptable ? If it's equeal to msn or messenger it's okay for me. My telephone bill is rissing like the sun. My familie oversea doesn't have any clue about computers so i like to setup this in linux so I can manage it.. Thanks you very much for your advice. Sjaak -- Dit bericht is gescand op virussen en gevaarlijke content en is veilig bevonden. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users