[asterisk-users] wip5000 crash AP
Good day all I have about 26 Hitachi WIP 5000 They all connect to the 4 Senao Long range AP's 11mb They all have the same ssi but 2 runs on channel 11 and 2 on channel 1 This way the roaming works well! We added a UPS and got POE injectors for each AP BUT..for some reason each now and the the AP's will crash, you can find a signal when you scan, and you can ping it, the only way to get it back up is to pull the power in and out I really don't know what else it can be and has giving up! Please help Altus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wip5000 roaming
Good day all I cant get my WIP 5000 to roam 100% I have 2 access points, different SSIs I make a config1 and config2 on the phone, each for the different SSIDs(A B) Im standing next to A and I walk to B, butthe phone does not want to change its signal to B, it still keeps the bad signal from A If I power A down, it will switch to B, if I switch A back on and go stand next to it, it will still keep Bs signal We got some wireless specialists in and they set up WDS for us, in other words, you add 1 SSID for both access point IT works for windows, but not for the phone! Can anyone help, or give a bit more explanation on the roaming settings on the webconfig Try RxLevel(-103~0) PreRoaming Enable RxLevel(-103~0) Try Over TxError Count(0~1) Try Over RxError Count(0~1) Level Diff Higher Than Curr Site(0~255) Use Refresh PreRoaming Enable PreRoaming On Association PreRoaming Mode PreRoaming Refresh Interval(0:Disable, 0~3600) Thanks Altus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] wip5000 roaming
Everything is working beside roaming Yes im using encryption, should I turn it off, or uses the same wep key, and same ssid Should I then also just add 1 config with 1 access point , not 2? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Thursday, November 09, 2006 8:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wip5000 roaming Disable WDS but set all the AP to the same channel and same SSID and then make sure they are connected to the same LAN (IE: no NAT on the AP). Are you using encryption? Something like: Try RxLevel -60 PreRoaming Enable RxLevel -75 Try over TxErrcnt 15 Try Over RxError Count 10 Play with the PreRoaming mode, see if it does help? It should however you could notice a drop in battery life. Would be a good place to start with your settings, adjust from there. I would like to hear your results with these phones, is everything working great besides the roaming? On 11/9/06, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I cant get my WIP 5000 to roam 100% I have 2 access points, different SSI's I make a config1 and config2 on the phone, each for the different SSID's(A B) Im standing next to A and I walk to B, butthe phone does not want to change its signal to B, it still keeps the bad signal from A If I power A down, it will switch to B, if I switch A back on and go stand next to it, it will still keep B's signal We got some wireless specialist's in and they set up WDS for us, in other words, you add 1 SSID for both access point IT works for windows, but not for the phone! Can anyone help, or give a bit more explanation on the roaming settings on the webconfig Try RxLevel(-103~0) PreRoaming Enable RxLevel(-103~0) Try Over TxError Count(0~1) Try Over RxError Count(0~1) Level Diff Higher Than Curr Site(0~255) Use Refresh PreRoaming Enable PreRoaming On Association PreRoaming Mode PreRoaming Refresh Interval(0:Disable, 0~3600) Thanks Altus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best gui
Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sipura 3000 Call waiting on the PSTN line
OK I have set the time and message Luki writes: Has anybody been able to get call waiting on the PSTN line? As far as I recall, you will only hear a tone in the audio stream when a second call comes in. The Sipura does not detect or handle it, but if you flash the line on the FXS interface after hearing the tone, the Sipura will forward the flash to the FXO interface and hence switch to the second call. I am positive this works when the call is picked up on the local FXS port but I am not sure if it also works when the call is picked up by a remote device. This is how I had set it up: PSTN - FXO on Sipura 3K - Asterisk (remote via DSL) - FXS on Sipura 3K - Phone The call would be re-invited in this case so no RTP traffic goes via DSL, only SIP traffic. Switching to second call with flash works in this scenario. Additionally I also allowed the call to be received by a remote device (RTP via DSL) but I am not sure if you can then use Call Waiting (never tried it). I don't think I'm expressing myself clearly here; if not, please ask. Or correct me if I'm wrong. Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how many oh323
Good day. I configured asterisk and oh323.Im using it as a sip-h323 convertor A call will come in to the asterisk box via IAX and be send to a quintum h323 gateway. in oh323 you can set the max in,out and simultaneous calls, Ive set them all to 100. Calls coming in via iax is alaw and then goes out h323 g729. It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing. Is there someone else with a setup like this.Is the problem on the asterisk side or the quintum Please help Thanks Altus ___ --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
Re: [Asterisk-Users] Sangoma and Digium same machine?
I have a Sangoma and a voicetronix openline 4 card. The trick is you had to uses asterisk cvs On Mon, 2005-09-26 at 13:07 -0400, William Lloyd wrote: Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill [EMAIL PROTECTED] ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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] cdr server
Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus ___ --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
Re: [Asterisk-Users] pri gateway
what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P # lsmod wct4xxp 106688 62 zaptel226820 129 wct4xxp # asterisk -r gw*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
Re: [Asterisk-Users] pri gateway
These are my configs for a sangoma 4 port connected to E1's in the UK loadzone = us loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 # card 0 - span 1 bchan=1-15,17-31 dchan=16 # card 0 - span 2 bchan=32-46,48-62 dchan=47 # card 0 - span 3 bchan=63-77,79-93 dchan=78 # card 0 - span 4 bchan=94-108,110-124 dchan=109 and zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown priindication = outofband usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes callgroup=1 pickupgroup=1 ; card 0 - span 1 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 1-15,17-31 ; card 0 - span 2 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 32-46,48-62 ; card 0 - span 3 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 63-77,79-93 ; card 0 - span 4 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 94-108,110-124 Maybe its your telco?? On Thu, 2005-09-08 at 15:23 +0300, Baris Simsek wrote: hi, my asterisk version is 1.0.9 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 it is comfortable with Turkish Telecom. i tried before and it works. /etc/asterisk/zapata.conf [channels] switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-15,17-31 Leds are lighting at start. When i run /etc/init.d/zaptel they go out. And i can see the modules are installed. and i see that, layer 1 is going up after zaptel. So i am sure there is no problem with drivers. I think it is connected to asterisk. any idea? thanks... altus wrote: what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P # lsmod wct4xxp 106688 62 zaptel226820 129 wct4xxp # asterisk -r gw*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
RE: [Asterisk-Users] queues
Hi So if I have this queues.conf [general] [default] [example_queue] music = default strategy = rrmemory context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 20 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = SIP/101 member = SIP/102 member = SIP/103 member = SIP/104 extensions.conf exten = 3,1,Playback(some_announce) exten = 3,2,Queue(example_queue|tT|||300) exten = 3,3,Dial(SIP/100) It will ring 104 for 20s,then 103 for 20s,then 102 for 20s and then 101 for 20s. It will keep on doing this for 300s then go the 100 If a second call comes it,it will start at 103 then 102 ens? Thanks for the help On Wed, 2005-09-07 at 08:07 +0200, Jens von Bülow wrote: Hi Altus, Try roundrobin with memory... snip Calls are distributed among the members handling a queue with one of several strategies, defined in queues.conf ringall: ring all available channels until one answers (default) roundrobin: take turns ringing each available interface leastrecent: ring interface which was least recently called by this queue fewestcalls: ring the one with fewest completed calls from this queue random: ring random interface rrmemory: round robin with memory, remember where we left off last ring pass /snip Also, as a rule of thumb, if you look at a call queues from the clients' perspective, a ringall strategy is what you have to do... (the others just can add huge delays in answering a call). Hope that Helps Jens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of altus Sent: 07 September 2005 07:57 AM To: asterisk Subject: [Asterisk-Users] queues Good day all I need some help with queues please. I know how to do a rounrobin in the queues.conf but I dont think its going to work in this situation Got got a IVR setup and option 3 is sales The sales people are 101,102,103,104 and the switchboard is 100 The trick comes is The 1st call for extension 3 goes to 101,but if 101 does not answer in 20 it goes to the switchboard,100 Then the second call of the day goes to 102,if not answer in 20s it goes to the switchboard,100 and so on and then just starts over again. Do I uses queues for this and then how?If I put it in a queues.conf and a roundroben,wont it then just try 101,and if not answer then 102 and if no answer 103...and so on? This is my queses.conf [general] [default] [example_queue] music = default strategy = roundrobin context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 20 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = SIP/101 member = SIP/102 member = SIP/103 member = SIP/104 and my extensions.conf exten = 3,1,Playback(some_announce) exten = 3,2,Queue(example_queue|tT|||20) exten = 3,3,Dial(SIP/100) h aph raph h Æ -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
Re: [Asterisk-Users] PRI in and out
I got the same setup,sort of I connected a single port sangoma to my pbx My ony problem is,when a call comes in and it gets transfered back out that it does not detect the hangup?So that channel keeps being open Any ideas why On Wed, 2005-09-07 at 01:40 -0600, Rich Adamson wrote: I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI - Asterisk side to the Asterisk - PBX side, or will a 4-port PRI card do the job? (I already have a spare one of these). The 4-port card will work just fine. In other words, can I use SPAN 1 as a timing source, then provide timing to the PBX connected to SPAN 2 of the same card? Yes. In fact, the 4-port card will be a slight advantage over two single port cards as all ports on the 4-port card will have their clocks in sync with your external timing source. Keep in mind that all T1/E1 spans having timing embedded in their transmit legs; you can't turn that off even if you tried. The clock timing source is always an engineering decision as to chosing which receive leg to use for clock sync. (Obviously, the span from the pstn would be your timing source and not the span to the pbx. If you already are using the PRI with the PBX, then no changes required on the PBX side for clock sync.) The config examples in zapata.conf and the wiki are good. Not much to configure really. You will probably want to focus more on options that your pstn provider can/will impact such as the number of digits to be sent from them to you, which channel is the d channel, the digits they expect from you for each call (whether prefixed with 1, 0 or whatever), etc. As sort of a side note, the 4-port card gives you another slight advantage from an ongoing support perspective. The third (or forth) port could be connected to a test asterisk box on which you can stage/test future asterisk code before moving it into the production box. Think about reserving a couple of DID numbers for the test box if you'll be using DID. ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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] queues
Good day all I need some help with queues please. I know how to do a rounrobin in the queues.conf but I dont think its going to work in this situation Got got a IVR setup and option 3 is sales The sales people are 101,102,103,104 and the switchboard is 100 The trick comes is The 1st call for extension 3 goes to 101,but if 101 does not answer in 20 it goes to the switchboard,100 Then the second call of the day goes to 102,if not answer in 20s it goes to the switchboard,100 and so on and then just starts over again. Do I uses queues for this and then how?If I put it in a queues.conf and a roundroben,wont it then just try 101,and if not answer then 102 and if no answer 103...and so on? This is my queses.conf [general] [default] [example_queue] music = default strategy = roundrobin context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 20 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = SIP/101 member = SIP/102 member = SIP/103 member = SIP/104 and my extensions.conf exten = 3,1,Playback(some_announce) exten = 3,2,Queue(example_queue|tT|||20) exten = 3,3,Dial(SIP/100) h aph raph h Æ -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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] Snom 360 problem
Good day all I have asterisk on a box with one network card I have a 2 companies setup on the system. To keep all apart I binded a different ip to the interface,i,o,w eth0 192.168.0.254 and eth0:1 192.168.1.254 And in sip.conf I took the bind setting out So each company's phones are on a different ip range,and all worked well So we decide to pull the snom190 out and exchange it with a snom360 This company is on the virtual interface(eth0:1) The 360 register and can make outgoing calls fine But when you try to make a call to it it does not work? I gives this error in the cli Forbidden - wrong password on authentication for INVITE to '301 sip:[EMAIL PROTECTED];tag=as3405ec0a But its 301 calling the snom360(user 310)??? BUT If I change the phone's ip and tell it to connect to eth0,not eth0:1 it works,same account settings same everything? The snom190 worked this way Any Ideas why? -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
Re: [Asterisk-Users] Snom 360 and hints
We schedule a reboot each night at 11/12 so it clears any errors,or hanging channels,in case for save keeping Im just not sure then with the 360 that it would keep the panel lights working. Got those problems your talking about,lost with the 220's On Fri, 2005-09-02 at 09:39 +0200, Remco Barende wrote: My experience with auto-rebooting schemes is that reliability doesn't improve. I also reported the non-registration of the Snom phones as a bug. On a few occasions I found that the phone lost registration, and rebooting or power cycling the phone didn't help (athorization failed at the * console) but re-entering the password in the phone did. Would this match your problem too? I guess it would be nice if we could make * log each password with which a SIP client tries to register (yes I am aware of the security implications of it). My guess is that the password disappears from the phone or it is corrupted. On Fri, 2 Sep 2005, altus wrote: SO if I do a reboot of the system each night at 12,it should be up and working again at 8 in the morning? On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should disappear is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
Re: [Asterisk-Users] Snom 360 and hints
IS there a way to make the phone reboot each day at a time? On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) First, ensure that the 360 has Filter Packets from Registrar turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do sip show subscriptions from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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
Re: [Asterisk-Users] Snom 360 and hints
SO if I do a reboot of the system each night at 12,it should be up and working again at 8 in the morning? On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should disappear is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --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] h323
Good day all How do I get h323 and video working? -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] h323
RFTW or RTFM On Wed, 2005-08-10 at 09:36 -0400, Mark Phillips wrote: With a liberal application of RFTW altus wrote: Good day all How do I get h323 and video working? -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] Asterisk to PSTN
What about a PRI/BRI solution We have a few with the voicetronix openline 4 cards and they work ok But the PRI solution work better.We have a 4 port sangoma configured like that.If I recall you set it to pri_net instead of pri_cpe and you use a cross cable On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote: hi folks. i'm planning to connect * to 120 POTS line. i've done some research on FXO cards but unfortunately most manufacturers only make 4 ports/card. the most i've found is 12 ports. so do i have to get 10 of these cards and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] h323
Good day all Im trying to get asterisk and oh323 to work I following the instruction on http://lists.digium.com/pipermail/asterisk-users/2005- January/081651.html It on fedora core 1,and I downloaded the lated dev. of asterisk Installation: tar -zxvf asterisk-oh323-0.7.1.tar.gz tar -zxvf pwlib-Janus_patch4-src-tar.gz tar -zxvf openh323-Janus_patch4-src-tar.gz cd pwlib ./configure make cd openh323 patch -p1 /root/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch (pach to openh323) cd openh323 ./configure make opt but at make opt I get this error g++: Internal error: Illegal instruction (program cc1plus) Please submit a full bug report. See URL:http://bugzilla.redhat.com/bugzilla for instructions. make[1]: *** [/root/openh323/lib/obj_linux_x86_r/h323rtp.o] Error 1 make[1]: Leaving directory `/root/openh323/src' make: *** [opt] Error 2 Can someone please help Thanks Altus -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] Speex QoS
isnt speedx just a codec used with sipiax? On Mon, 2005-08-08 at 08:12 -0400, Adam Robins wrote: Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] h323
Good day all Can I register asterisk as a h323 client,like in sip where you have register = -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] h323
What is the difference? Is it like register and registrar ? If I make asterisk like a server and clients connect to it,is it a gatekway? And if I call another gateway its a gatekeeper ? On Thu, 2005-08-04 at 11:14 -0400, Juan Salas wrote: From wiki... (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. As I look, asterisk didn't act like gatekeeper. JS. Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -Mensaje original- De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 10:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ 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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] qozap junghanns errors
Only one card:-) This is the second time I had it on a Intel board Even junghanns does not know about it Giorgio Incantalupo wrote: Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ 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
Re: [Asterisk-Users] qozap junghanns errors
Giorgio Incantalupo wrote: Thanks Will have a look Hi Altus, sorry about it. Have you tried to disable all you don't need on your server, for example parallel ports, serial ports, usb ports, etc?? Have you checked with cat /proc/interrups ?? Maybe your card share some interupt with other cards (eth0 for example). We are using Dell PCs but they do not let us to choose how to set interrupts, maybe your PC can. I'm sorry I cannot be more exaustive but this kind of problem is very hard to solve. Giorgio. Altus Snyman wrote: Only one card:-) This is the second time I had it on a Intel board Even junghanns does not know about it Giorgio Incantalupo wrote: Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ 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 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] Voice mailbox on the fly?
Why not exten = 123,1,BackGround(whatIsthe6Digets) exten = 123456,1,Voicemail(u123456) Jim Archer wrote: Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? 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 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] Zaptel error: Unable to create channel op type 'Zap'
I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, 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] qozap junghanns errors
Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ 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] Junghanns quadBRI on Dell PowerEdge
I had the same problems with a 4 port junghanns and a 4 por wcfxs I took the junghanns out and added it into a new box and all was ok So ether it was because the 2 cards was in together or it was the motherboard? U using the latest driver and asterisk? On Wed, 2005-07-20 at 11:31 +0200, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Loading qozap driver: Jul 19 17:15:55 ustredna kernel: qozap: no version for zt_receive found: kernel tainted. Jul 19 17:15:55 ustredna kernel: qozap: Junghanns.NET quadBRI card configured at mem 0xf8836000 IRQ 77 HZ 1000 CardID 0 Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Running ztcfg: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel map: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel 01: Individual Clear channel (Default) (Slaves: 01) Jul 19 17:15:56 ustredna ztcfg: Channel 02: Individual Clear channel (Default) (Slaves: 02) Jul 19 17:15:56 ustredna ztcfg: Channel 03: D-channel (Default) (Slaves: 03) Jul 19 17:15:56 ustredna ztcfg: Channel 04: Individual Clear channel (Default) (Slaves: 04) Jul 19 17:15:56 ustredna ztcfg: Channel 05: Individual Clear channel (Default) (Slaves: 05) Jul 19 17:15:56 ustredna ztcfg: Channel 06: D-channel (Default) (Slaves: 06) Jul 19 17:15:56 ustredna ztcfg: Channel 07: Individual Clear channel (Default) (Slaves: 07) Jul 19 17:15:56 ustredna ztcfg: Channel 08: Individual Clear channel (Default) (Slaves: 08) Jul 19 17:15:56 ustredna ztcfg: Channel 09: D-channel (Default) (Slaves: 09) Jul 19 17:15:56 ustredna ztcfg: Channel 10: Individual Clear channel (Default) (Slaves: 10) Jul 19 17:15:56 ustredna ztcfg: Channel 11: Individual Clear channel (Default) (Slaves: 11) Jul 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] Junghanns quadBRI on Dell PowerEdge
The 1ste pc I tried it on was on a expensive intel board and the second one that worked was on some cheap name board Ill say incompatibility ? Yes, I do use latest bri-stuff package (asterisk 1.0.9 incl) Any ideas? - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com altus wrote: I had the same problems with a 4 port junghanns and a 4 por wcfxs I took the junghanns out and added it into a new box and all was ok So ether it was because the 2 cards was in together or it was the motherboard? U using the latest driver and asterisk? On Wed, 2005-07-20 at 11:31 +0200, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Loading qozap driver: Jul 19 17:15:55 ustredna kernel: qozap: no version for zt_receive found: kernel tainted. Jul 19 17:15:55 ustredna kernel: qozap: Junghanns.NET quadBRI card configured at mem 0xf8836000 IRQ 77 HZ 1000 CardID 0 Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Running ztcfg: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel map: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel 01: Individual Clear channel (Default) (Slaves: 01) Jul 19 17:15:56 ustredna ztcfg: Channel 02: Individual Clear channel (Default) (Slaves: 02) Jul 19 17:15:56 ustredna ztcfg: Channel 03: D-channel (Default) (Slaves: 03) Jul 19 17:15:56 ustredna ztcfg: Channel 04: Individual Clear channel (Default) (Slaves: 04) Jul 19 17:15:56 ustredna ztcfg: Channel 05: Individual Clear channel (Default) (Slaves: 05) Jul 19 17:15:56 ustredna ztcfg: Channel 06: D-channel (Default) (Slaves: 06) Jul 19 17:15:56 ustredna ztcfg: Channel 07: Individual Clear channel (Default) (Slaves: 07) Jul 19 17:15:56 ustredna ztcfg: Channel 08: Individual Clear channel (Default) (Slaves: 08) Jul 19 17:15:56 ustredna ztcfg: Channel 09: D-channel (Default) (Slaves: 09) Jul 19 17:15:56 ustredna ztcfg: Channel 10: Individual Clear channel (Default) (Slaves: 10) Jul 19 17:15:56 ustredna ztcfg: Channel 11: Individual Clear channel (Default) (Slaves: 11) Jul 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.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 mailing list Asterisk-Users
[Asterisk-Users] asterisk number of calls
Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] quadBRI form junghanns.net
Got a few and 8line one running good,got some compatibility problems with some mother boards once but that was it On Wed, 2005-07-06 at 16:08 -0300, Bartosz Jozwiak wrote: Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. Thank you in advance Bartosz ___ 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 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] UK asterisk
Good day all Im looking for someone in the UK that knows asterisk and thats willing to do a quick job for us,its in at tele city -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ 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] Asterisk and Sirrix PCI4S0 echo cancellation
On the subject of this in you /var/log/messages do you get errors like this Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user root by (uid=0) Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0 Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3 Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3 Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): ipac_handle_interrupt_icd: error, RSTAD = 0x1e not ok! Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3 email this guy,he wrote a patch to bring down the volume [EMAIL PROTECTED] On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote: Hi guys, I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS- HEAD (20050614). Something I've come across is that with 'echocancel = yes' in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in fact that the echo distorts. To remedy this I've set 'echocancel = no' and disabled the echo cancellation. With the echo cancellation disabled there is still an echo but it is much softer. Any ideas on how I can turn on the echo cancellation again without having the very loud echo back ? Is there some way I could perhaps drop the TX volume out of the Sirrix card ? Perhaps this would help ? Thanks in advance. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. ___ 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
Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation
No,sometimes i get a watery sound,like when you speak under water I turned echo off,do you have the latest driver.A new version came out on the 16th Altus On Tue, 2005-06-21 at 10:17 +0200, David Wilson wrote: Hi Altus, Thanks for your reply. Yes I do get those errors. Any ideas what causes them ? email this guy,he wrote a patch to bring down the volume Thanks I've been chatting with Steve already. For some reason the patch does not seem to be working with my newer version of the Sirrix driverseither that or I'm doing something wrong. :) Did you have to modify the patch in any way or did you just apply it 'as is' ? I will keep in touch to let you know the outcome. Many thanks. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: altus [EMAIL PROTECTED] To: asterisk asterisk-users@lists.digium.com Sent: Tuesday, June 21, 2005 9:52 AM Subject: Re: [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation On the subject of this in you /var/log/messages do you get errors like this Jun 21 09:38:45 pbxct sshd(pam_unix)[1993]: session opened for user root by (uid=0) Jun 21 09:39:22 pbxct kernel: 00: Unsetting old PCM master PFIC 0 Jun 21 09:39:22 pbxct kernel: 00: Setting PCM master to PFIC 0 IPAC3 Jun 21 09:39:22 pbxct kernel: Slip detected on IPAC3 Jun 21 09:40:02 pbxct kernel: sirrix ipac (3): ipac_handle_interrupt_icd: error, RSTAD = 0x1e not ok! Jun 21 09:40:07 pbxct kernel: Slip detected on IPAC3 email this guy,he wrote a patch to bring down the volume [EMAIL PROTECTED] On Tue, 2005-06-21 at 09:37 +0200, David Wilson wrote: Hi guys, I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS- HEAD (20050614). Something I've come across is that with 'echocancel = yes' in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in fact that the echo distorts. To remedy this I've set 'echocancel = no' and disabled the echo cancellation. With the echo cancellation disabled there is still an echo but it is much softer. Any ideas on how I can turn on the echo cancellation again without having the very loud echo back ? Is there some way I could perhaps drop the TX volume out of the Sirrix card ? Perhaps this would help ? Thanks in advance. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. ___ 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 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] ipswitchboard
Good day all Im trying to download ipswitchboard but the webpage does not seem to work? Can someone maybe put it somewhere,and the .NET thing you must install with it,please Or is there a different link to http://ipswitchboard.thorben.dk/ Please Thanks ALtus ___ 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] Dial more then 9 digits
no I can? how is your dialout rules ? I have a client where you have to dial a 4 digit pin and then the rest of the number I simply have a exten = _1234.,1,Dail... On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote: Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob. ___ 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
RE: [Asterisk-Users] Dial more then 9 digits
The
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
sys any 9 digit number
try _X.,1
On Wed, 2005-06-15 at 13:23 +0200, Bohuslav Coufal wrote:
my exten
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[default]
; If the number dialed by the calling party was 2000, then
; Dial the user 2000 via the SIP channel driver. Let the number
; ring for 20 seconds, and if no answer, proceed to priority 2.
; If the number gives a busy result, then jump to priority 102
;exten = s,1,Dial(SIP/${EXTEN})
;exten = s,1,Dial(SIP/7406100)
exten = 7406100,1,Dial(SIP/7406100)
exten = 7406101,1,Dial(H323/[EMAIL PROTECTED])
exten = 7406105,1,Dial(SIP/7406105)
exten = 7406106,1,Dial(SIP/7406106)
exten = 7406200,1,Dial(SIP/7406200)
exten = _74068XX,1,Dial(H323/[EMAIL PROTECTED])
exten = _OO.,1,Dial(H323/[EMAIL PROTECTED])
exten = _X,1,Dial(H323/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of altus
Sent: Wednesday, June 15, 2005 12:31 PM
To: asterisk
Subject: Re: [Asterisk-Users] Dial more then 9 digits
no
I can?
how is your dialout rules ?
I have a client where you have to dial a 4 digit pin and then the rest
of the number
I simply have a
exten = _1234.,1,Dail...
On Wed, 2005-06-15 at 12:20 +0200, Bohuslav Coufal wrote:
Could you kick me, I can't dial more then 9 digits. Is anyone some
default length of extensions or dialed number.
Thanks,
Bob.
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[Asterisk-Users] sirrix NT mode
Good day all Is there someone who's got a sirrix 4 port working in NT mode I got one working good in TE mode. Apparently I must add 8 jumpers in make the cross cable a straight cable But what about the sirrix.conf? Do I just change the mode from TE to NT? Please Help or advice? Thanks Altus ___ 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] lost g729 lic
Good day all We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please Help ___ 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] lost g729 lic
Thank up very much for the response Its appreciated and it will help me allot I hope u have a nice Monday or is it Friday? ALtus (the early-morning BOER!) On Fri, 2005-06-10 at 06:05 -0400, Andrew Kohlsmith wrote: On Friday 10 June 2005 05:09, altus wrote: We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please do a modicum of research, hell even contact the people you got the licenses from (i.e. [EMAIL PROTECTED]). This kind of question is insulting to this entire list because it shows a total lack of resepect for everyone on it. We enjoy helping others, but at the same time there is a basic level of research which is requested in this social contract, and which you haven't displayed. -A. (the early-morning list nazi, apparently) ___ 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
Re: [Asterisk-Users] 4 port BRI options ?
Or the sirrix,I think its the cheapest and there was lots of dev. on the drivers On Fri, 2005-06-03 at 04:49 -0700, Nardis Dome wrote: Hi, Eicon Diva 4BRI Card and chan_capi. --- Brett, Gary [EMAIL PROTECTED] wrote: Hi there I am in the UK.. and am running latest asterisk on FC1 (2.4 kernel). I would like to know what the best option is for a 4 port BRI card. I notice Digium don't provide one.. I have heard the Junghanns do one...but are there others ?? Is the Junghanns card reliable/stable with good sound quality ?? I notice it is very expensive in a per port comparison with the Digium cards hence why I am also looking for alternative cards Your experiences would be greatly appreciated Gary ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 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] sox
Good day all I remember some time ago I tried recording on asterisk But it did not work because the sox app was broken and by downloading a older one it worked Now things have come and go and version change What sox version will work with asterisk 1.0.7 Thanks Altus ___ 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] monitoring
What I done is On the asterisk box I out this in crontab for each min. ps -ax | grep -v grep | grep -i asterisk || echo asterisk not running |mail -s asterisk on XDFS is down [EMAIL PROTECTED] this will mail me if asterisk isnt running We have a cell provider that offers email to sms so I can just add a cc option Also running on the box,each morning at 7am it does asterisk -rx 'sip show peers' /tmp/file sleep 5 cat /tmp/file | mail -s Check sip users @ company [EMAIL PROTECTED] Then On the asterisk box I enable iax2,so that port 4569(tcp) is open Then I get a app called udpping(got it off the wiki) This will do a udp ping on a ip for port 4569 So On a different box,I do in the crontab for each 5min /bin/udpping 1.2.3.4 || echo no connection to the box | mail -s no connection [EMAIL PROTECTED] Or you could just use nagios And another app is called swat,does a tailgrep on log files (/var/log/asterisk/messages) and greps for words and mails you if it finds it But this works for me Let me know how it goes ALtus [EMAIL PROTECTED] On Tue, 2005-05-31 at 18:11 -0500, jltaylor wrote: Has anyone done any scripts (or something else) to notify if something goes down? Example: Asterisk_1 is peered to Asterisk_2 Asterisk_1 has qualify=yes Asterisk_1 notices that Asterisk_2 is not responding Asterisk_1 sends email to cell phone Asterisk_2 peer down James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ 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] deadlock
All out of the blue I get these errors? Any Ideas why Please help May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:33 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:39 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:41 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:45 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:51 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retrie ___ 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] capi.conf
Good day all I need some help What is the device= in capi.conf How will it look for a 4 port card? I have a section for each msn...but how do I tell witch msn is witch port because in the extensions.conf I dail [EMAIL PROTECTED] Please give me a example of more than one file Thanks Altus ___ 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] chan_capi error2
On
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
it tells u if u use the cvs as of april you need a patch
I have bot
I tried and it compiled and there is no errors in asterisk startup
What did u change in the capi.conf file?Is it ok if I just change the
context
Thanks
Altus
On Fri, 2005-05-20 at 13:35, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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Re: [Asterisk-Users] chan_capi error2
A fix for what?
I think the patch in that link is broken because I had to take out a lot
of end of lines
Dont you maybe have a working patch
Thanks for the help
Just a question about the conf file
msn and incomingmsn
What is the difference
is msn what you uses when you with the Dial command and incomingmsn is
what is send to extensions.conf?
Thanks again
Altus
On Fri, 2005-05-20 at 14:32, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
On
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
it tells u if u use the cvs as of april you need a patch
I have bot
I tried and it compiled and there is no errors in asterisk startup
I don't think the patch is necessary with your version, but it contains a
fix.
I don't know what the problem with your compilation is, maybe you can
provide more output.
What did u change in the capi.conf file?Is it ok if I just change the
context
Sorry, but what do you mean? You need to setup up a capi.conf according to
your ISDN lines/numbers.
Armin
On Fri, 2005-05-20 at 13:35, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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[Asterisk-Users] chan_capi patch eicon
Good day all Im trying a eicon 4bri card On fedora core 1 I installed the rpm,lsmod says the driver is working I then installed asterisk 1.0.7 I then download chan_capi 0.3.5 But now it says I should patch it for asterisk So I got the patch..fixed it And did a make and it gives a lot of syntax errors Please Help Thanks Altus ___ 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] eicon fdc3
Good day all Did anyone get the eicon 4 bri working with asterisk and fedora core 3 Please Thanks Altus ___ 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] fdc3 no gsm
Good day all I installed Fedora core3 I also installed mpg123 0.59r but asterisk does not want to play anything..on 2 of my server No BAckgroung,Voicemail..nothing Never had this before In the cli it shows its playing it But nothing happens? Please Help ___ 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] 2 servers via PRI
Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to pri_net...this cant be all? And the cable pin1 -- pin4 pin2 -- pin5 pin3 -- pin6 pin4 -- pin1 pin5 -- pin2 pin6 -- pin3 pin5 -- pin8 pin8 -- pin7 Please Help and advice Thanks Altus ___ 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] cdr!
Good day all I installed asterisk-addons and now its logging nicely in my database But I want it to log in my usual log csv as well Please Let me know Thanks Altus ___ 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] qozap(!) problem
Same..8a On Mon, 2005-05-09 at 17:12, Eugenio De Vena wrote: Which version of * and bristuff did you install, I had bristuff-0.2.0-RC8a and now I am trying bristuff-0.2.0-RC8c - Original Message - From: Altus Snyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2005 3:15 PM Subject: Re: [Asterisk-Users] qozap(!) problem Ya well let me know when u solved this We have the same thing Do you have any other cards in with it We have a diguim fxs/fxo card in so maybe its a error with working together Anyway Let me know when you get a fix for it because no one seems to know(or check their /var/log/messages) This lets my asterisk hang at lest one daily and I needed to schedule regular reboots On Mon, 2005-05-09 at 14:45, Eugenio De Vena wrote: As I said before, I can not get help from junghanns, so I ask the list. I installed * version 1.0.7 bristuffed latest version and this solves the music on hold problem. But this introduces a new problem that I did not have before. Every 1 second pops up the message: May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 8 z1 64 z2 40 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 26 z2 0 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 1 z1 32 z2 15 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 63 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 2 z1 34 z2 16 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 33 z2 7 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 104 z2 77 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 4 z1 77 z2 57 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 3 z1 61 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 31 z2 5 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 10 z2 112 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 100 z2 74 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 62 z2 35 there are no IRQ conflicts ( checked with lspci -v) and everything works. What does this message mean? Thanks for any help Eugenio ___ 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
Re: [Asterisk-Users] Stun codec
I uses to have this when I enabled stun and did not need it On Tue, 2005-05-10 at 16:55, Ronald Wiplinger wrote: I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the order: g729 g711u g711a Any ideas, why the user can hear me, but I cannot hear him (stun) while the other user without stun has no problem. bye Ronald ___ 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] asterisk-addon
Good day all I downloaded asterisk-addons to try and make asterisk log in the sql db but when I make a make install i get this error cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Please help ___ 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] transfer queues agents
Good day all This is what i got off the net about queues and agents Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call. We have a snome 220 that does consultative transfer..with the buttons on the phone Does this mean I wont be able to do this? Please Help and andvice ___ 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] sangoma fdc 3?
How well does the sangoma cards work with fedora core 3 Im doing the research on what hardware/os I need to use Please help and advice ___ 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] qozap(!) problem
Ya well let me know when u solved this We have the same thing Do you have any other cards in with it We have a diguim fxs/fxo card in so maybe its a error with working together Anyway Let me know when you get a fix for it because no one seems to know(or check their /var/log/messages) This lets my asterisk hang at lest one daily and I needed to schedule regular reboots On Mon, 2005-05-09 at 14:45, Eugenio De Vena wrote: As I said before, I can not get help from junghanns, so I ask the list. I installed * version 1.0.7 bristuffed latest version and this solves the music on hold problem. But this introduces a new problem that I did not have before. Every 1 second pops up the message: May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 8 z1 64 z2 40 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 26 z2 0 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 1 z1 32 z2 15 May 9 13:31:54 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 63 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 2 z1 34 z2 16 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 33 z2 7 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 104 z2 77 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 4 z1 77 z2 57 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 3 z1 61 z2 42 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 31 z2 5 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 10 z2 112 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 10 z1 100 z2 74 May 9 13:31:55 coscosmia kernel: qozap: dropped audio card 1 cardid 0 bytes 11 z1 62 z2 35 there are no IRQ conflicts ( checked with lspci -v) and everything works. What does this message mean? Thanks for any help Eugenio ___ 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] qozap message error
Good day all with the laster driver and latest drive asterisk I get these errors Please help May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1 71 z2 36 May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 21 z1 30 z2 121 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 20 z1 21 z2 113 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 21 z1 86 z2 49 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1 63 z2 28 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 21 z1 53 z2 16 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 20 z1 29 z2 121 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 22 z1 5 z2 95 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 20 z1 106 z2 70 May 3 17:43:15 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 20 z1 54 z2 18 ___ 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] bri error
Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ 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] bri error
if I do a zttool it shows TE mode On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ 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
RE: [Asterisk-Users] bri error
and I have signalling = bri_cpe_ptmp On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ 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] bri cli error
Good day all I get this error in my cli chan_zap.c:7407 zt_pri_error: PRI: !! Got a UA, but i'm in state 0 I have a 4 port Junghannes card connect with 2 bri isdn lines It keeps on dropping calls and giving errors Please help and advice Thanks ALtus ___ 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] security
Good day all I want to put a asterisk server on a public ip and allow any,registered sip and iax connection What security risks are there and how can I secure my pabx One thing I want to know is how do I make it that anyone can call a extension at my box but not make a call out. i.o.w how do I call [EMAIL PROTECTED] and how do I make it that it cant call [EMAIL PROTECTED] Please help me with these question Thanks Altus ___ 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] analog gsm router
Good day all I have a analog gsm router and a 4 port bri card:-) How do I get the gsm router to work with asterisk I tried adding a voicetronix card but the 2 cards doen not seem to work together,it gives a unresolved symbols error when starting up Any Ideas Please Can you add 2 zaptel device,different ones? Like the Junghannes and a diguim analog card? Please help and advice Thanks ALtus ___ 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] hangs pc
Good day all I installed asterisk on a pc with redhat 9 and a 4port bri eachtime a call comes in,iax,sip,pstn it just hangs the pc Top shows 75% of the cpu goes to asterisk? Any Idea why? Please Help ___ 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] qos test
Good day all I'm looking for a type of QOS test tool(software) I want to test if a link is good enough for voip and test witch ones will be the best..ens any ideas ___ 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] pbx to asterisk
Good day all I just want to know if someone tried this and with out any hassles What I want to do is take 4 extension(analog) of a current,old,pabx unit and put them into a asterisk server with a 4port analog card,like the voicetronix openline4 card. (PSTN)(old PABX)---===(4 ports asterisk) Please Help Altus ___ 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] voicetronix bri
Good day all Will a voicetronix openline 4 card work with a 4port BRI card? Please HElp/advice ___ 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] voicetronix bri
Voicetronix will only be used for the gsm cell router and BRI for outgoing-incoming calls On Thu, 2005-04-14 at 11:26, Michael Bielicki wrote: In what sense ? voicetronix is analog BRI is ISDN digital On 4/14/05, Altus Snyman [EMAIL PROTECTED] wrote: Good day all Will a voicetronix openline 4 card work with a 4port BRI card? Please HElp/advice ___ 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] voicetronix dtmf
Good day all I got the latest cvs asterisk But when making a call out threw the voicetronix openline4 card the dtmf doens not work I got this in vpb.conf ecsuppthres = 4096 indication = 1 dtmfidd = 3000 ast-dtmf-det=1 relaxdtmf=1 break-for-dtmf=yes Please help Thanks Altus ___ 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] fedora 3
Thanks for the trouble n Wed, 2005-04-06 at 15:00, iMRAN wrote: Hi, I`ve installed on FC-3 last month and its working gr8... no probs so far Imran On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have a Fedora core 3 installation Is there any hassles with asterisk? Thanks Altus ___ 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] Planet VIP 450
Good day all Did someone get the planet VIP 450 working Thanks Altus ___ 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] snom220
Good day all I'm looking for someone with good knowledge of the way the snom220 transfer I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to 200 1st and the transfer B to 200. Please Help Thanks Altus ___ 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] snom220
Does Call join on Xfer (2 calls) be on or off? Thanks On Fri, 2005-04-01 at 04:29, Damon Estep wrote: I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to 200 1st and the transfer B to 200. Easy. Park the call, call B and talk to him and tell him where the call is parked This applies to the SNOM 190 which should be the same as the 220 Make sure the break key = off in the snom web based setup utility, after this is off the transfer key will bridge the last two active calls. So you are on a call on line 1, line 2 rings You answer line 2 by pressing the flashing button (hold key not needed, it is automatic). Press the third line button (again, hold is automatic), talk to the third party, and press the transfer key when ready, the line 2 and line 3 will de bridged and they will disappear from the phone. Press line 1 to resume the original call. ___ 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] sox
Good day all I previously tried the Monitor app with sox but it did not work and according to the list it was because of a broken version What are a good and working version for the latest asterisk Thanks altus ___ 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] Fax and Voice
google asterisk fax On Thu, 2005-03-24 at 11:53, Guy Decarpentrie wrote: Hi all, Is * able to do the difference between Fax and voice, and then adapt the treatment of the call ? An example ? Thx ___ 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
Re: [Asterisk-Users] Fax and Voice
exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit : google asterisk fax Well, i know how to receive and mail a fax, now i want to know how to detect if the call is a fax or a voice call, and reroute the call if it's a voicecall, and mail the fax if it's one. I think you need to follow the original directions: go to google, search for asterisk fax The very first hit tells you exactly what you want: Fax Detection with IAX and SIP If you are trying to detect faxes over IAX, SIP, or for that matter any type of channels, Newman Telecom has created NVFaxDetect and updated BackgroundDetect? as NVBackgroundDetect for that purpose. We have had near perfect results on decent IAX connections using ULAW/ALAW. Fax detection utilizes Asterisk DSP and works in the same way once detected, faxes are sent to the fax extension. See Asterisk fax for example fax detection scripts. and has links to another part of the wiki where examples are given. Ok thanks to all, i've to wake-up... ___ 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
Re: [Asterisk-Users] Fax and Voice
sorry exten = fax,1,Dail On Thu, 2005-03-24 at 12:53, Altus Snyman wrote: exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit : google asterisk fax Well, i know how to receive and mail a fax, now i want to know how to detect if the call is a fax or a voice call, and reroute the call if it's a voicecall, and mail the fax if it's one. I think you need to follow the original directions: go to google, search for asterisk fax The very first hit tells you exactly what you want: Fax Detection with IAX and SIP If you are trying to detect faxes over IAX, SIP, or for that matter any type of channels, Newman Telecom has created NVFaxDetect and updated BackgroundDetect? as NVBackgroundDetect for that purpose. We have had near perfect results on decent IAX connections using ULAW/ALAW. Fax detection utilizes Asterisk DSP and works in the same way once detected, faxes are sent to the fax extension. See Asterisk fax for example fax detection scripts. and has links to another part of the wiki where examples are given. Ok thanks to all, i've to wake-up... ___ 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 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] snom220 problem
Good day all I have a snom 220 with the extra keypad When more than one call comes in none of the extra lines on the phone lights up or anything.You hear the beep in you ear but no way of picking it up.I tied 4 different firmware versions.On was a very old one,with actually worked but is gave echo and got slow and hanged up. So the button are ok I just think that maybe there are some type of setting on newer version that needs to be disabled or something Please Let me know ___ 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] snom 220 version
Good day all What is a good stable snom 220 firmware version. Thanks Altus ___ 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] Using Codec G-726
had the same thin with 729 I had to go disallow=all allow=g279 On Thu, 2005-03-17 at 16:37, Matt wrote: Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work... ___ 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] snom 220 busy all the time
Good day all We have a snom 220 that for some reason keeps on giving this message Got SIP response 486 Busy Here back from 192.168.21.222 even though there is no active calls to it and there are 2 accounts set on the phone? Please Help and advice Thanks Altus ___ 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] from sip to asterisk to h323..how
Goo day all This is our setup Client phone--(SIP)--asterisk server---SIP/IAX---asterisk--- -- goes out to international server running sip/iax But now I want to dial out to H323 server? I.O.W I want asterisk to act as a H323 client that will rout some calls out to a H323 server.How do I do this an can asterisk eve do this I had a quick look on the net and only saw that asterisk can be a h323 server not client. Please Help Thanks Altus ___ 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] iax,trunking,zap
Good day all Why do I need a Zaptel card to do trunking in IAX?? What if I only had a voice/iax router? Is there a way around this? Thanks Altus ___ 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] IAX+G729a
Good day We are going to add 6 channels of G729a to our asterisk server running iax between them I have a few question about the hole license thing. In iax.conf do i allow g729 or g729a?What's the difference? This license is for 2 servers,i.o.w 3 per server.How many calls does this give us? For example if server A calls server B does it uses 1 license,server A's license, or does it use 2,1 for each server. If all the licensed channels are used,how do I let it know to uses the next available codec.Currently it give a error about running out of codec! Please help and Advice Thanks Altus ___ 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] snom220 *8 hangup
Good day all We have a snom 220 set as a switchboard phone I also configured *8 so that if the operator is somewhere else and it rings she can just go *8 on the nearest phone,Grandstrams bt-100 and snom 190.But If she does this she only speaks for about 30s and it will cut off the caller? Any ideas Altus ___ 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] hylafax
Good day all Can hylafax work with asterisk..and how I'm trying to find a way to send a fax over my E1 connection Please Help Altus ___ 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] Difference between E1 and PRI
PRI comes in 2versions E1 European and T1 US E1 30 channels T1 23 channels On Wed, 2005-02-23 at 14:15, Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true? ___ 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
Re: [Asterisk-Users] does asterisk support menus?
Yes Application Background() On Tue, 2005-02-22 at 14:35, Muhammad Muzzamil Luqman wrote: Whenever some call comes in i want it to be automatically picked up and then it plays some message Welcome to xyz, Press 1 for sales and 2 for support and then it takes it to the particular extension of sales/support. can i achieve this thing using asterisk? Kindest Muhammad Muzzamil Luqman __ ___ 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] send fax with pri
HI all What is the best to send a fax with a PRO. I got it working on the receiving and e-mailing it.How do I send one Thanks Altus ___ 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] route outgoing call
Good day all
I registered at a few sip server in different countries
Now I want to route outgoing calls for that country threw that sip
server and all the others there my own pstn,ZAP card.I already
registered asterisk with them.
How would my extensions.conf look.This is what I have but no matter what
it still goes there my server.We dial 9+countrycode to get to that
country.So on the pbx 0944... will go to the UK.
Here is what I have.Please help me correct this
ignorepat = 0
;UK
exten = _0944.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],50)
exten = _0944.,2,Congestion
;USA
.
.
;--Germany
.
.
;--All other
exten = _0.,1,Dial(Zap/1/${EXTEN:1})
exten = _0.,2,Dial(Zap/2/${EXTEN:1})
exten = _0.,3,Congestion
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[Asterisk-Users] Sangoma A101
Good day all Is there any difference in the sangoma zaptel.conf and zapata.conf then other cards Thanks altus ___ 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] Sangoma A104 - D-Channel problem
While on sangoma We are getting a samngom pri?Is there any driver I need to install,how does it work,like a Zaptel card. Any doc Please Let me know altus On Fri, 2005-02-18 at 11:06, Kumak wrote: On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote: upgrade to the following wanpipe and also upgrade the firmware o the crd (it's included in the wanpipe softwaare) ftp://ftp.sangoma.com/linux/custom/2.3.2/wanpipe-beta5g-2.3.2.tgz I did it before asking on the list. I have firmware ver8 on card and wanpipe-beta5g-2.3.2 but problem still exists. Here is wanpipe1.conf from wancfg [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 10 PCIBUS = 0 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = YES ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO ___ 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] asterisk qualified
Good day all Is there any time of VOIP/SIP/asterisk qualifications or certificates? Thanks Altus ___ 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] h323
Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice ___ 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] spandsp asterisk 3/5
Good day all I want to know with version of spandsp works well with ether asterisk 1.0.3 or 1.0.5 Thanks Altus ___ 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 in New-Zealand
Good day all Anyone doing asterisk in New-Zealand? ___ 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
