[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] 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
[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] 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
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
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
Re: [Asterisk-Users] Asterisk in Singapore.
I can get you a good deal if you import the from South-Africa..Let me know.Altus On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote: In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta Jonathan __ ___ 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 problem
Thanks Will have a look On Fri, 2005-02-11 at 09:59, Edin Kozo wrote: Hi Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Hope that helps you --- Altus Snyman [EMAIL PROTECTED] escribió: Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore 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 __ Renovamos el Correo Yahoo!: ¡250 MB GRATIS! Nuevos servicios, más seguridad http://correo.yahoo.es ___ 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] Cisco7960/SCCP Transfer Help?
If you select more there Trnsfer and BlndXfer will be displayed BlndXfer for Blind transfer Trnsfer for Confirm transfer This is on 7960 On Thu, 2005-02-10 at 15:09, [EMAIL PROTECTED] wrote: I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the Transfer softkey does not show up - as a matter of fact only 2 softkeys show up (redial something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are active. What am I missing as far as getting the Transfer button to show up on my SCCP phone? Additionally, the # does not work when talking on an outside line to do a transfer that way; it only works when talking to another internal phone I've intercommed. Help would be very much appreciated :-). Thanks, Bruce -- Bruce M. Himebaugh Himebaugh Consulting, Inc. 330/493-9700 http://www.hcd.net Computer consulting, software/web development systems integration CanNet Internet Services, Inc. 330/484-2260 http://www.cannet.com Providers of World-Wide Connectivity Get Connected ... Stay Connected! ___ 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 problem
Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore 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] limit iax calls
Good day all We have 2 asterisk servers,connected with iax2 and the phone via SIP They dont have a very big line so I want to restrict the call limet to 3 iax2 calls at a time,and for instance it the 4th call is made it will say something like all lines are being use try later 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] How to xfer calls or is my setup wrong?
What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss... The call comes in from an external number via IAX2 (0870xxx) which I can answer on any of the ringing extensions no problem. But when I need to xfer that call I am more or less stuck. I have read various posts and something about *8# ? seemed to partially work one on the grandstream but I haven't been able to reproduce that... The CLI doesn't show anything odd... 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 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
Good day all I have a asterisk installation,1.0.3, and spandsp. I got asterisk working,I edited the make file myself. Now when I receive a fax I only get half a page or nothing any Ideas why Please let me know 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] bri dropping calls
Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? 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] bri dropping calls
O did not have a look at it yet,I got the one from a week ago,how is aterisk 1.0.5? On Wed, 2005-02-09 at 08:04, Michael Bielicki wrote: hmmm the latest bristuff uses asterisk 1.0.5 so it can't be laast, can it ? cheers Michael On Wed, 09 Feb 2005 07:24:34 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? 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 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] sip_notify.conf
Good day all What is the file sip_notify.conf for 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] bri dropping calls
Where do you get this new version of bristuff,I had a look on the webpage and there's only RC3 On Wed, 2005-02-09 at 08:58, Peer Oliver Schmidt wrote: Altus Snyman wrote: We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Are you sure the call get dropped? We have a similar problem, but the call does not get dropped, but stays silent for a couple of seconds. If both parties don't hangup, they will be able to continue the conservation. (And yes, the latest to get is bristuff_0.0.2RC5 [RC6 seems to be for quadbri and octobri cards, only]) ___ 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] warning message
Good day all.I get the warning message on my system,this is for a snom 220,it repeats this message a few times,please help Feb 8 09:29:26 WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Non-critical Request) Is there a page that describes all asterisk's error and warning messages? 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] snom soft phone
Did you try 00 That is what it is on the 220 On Tue, 2005-02-08 at 09:36, Paradise Dove wrote: what is the password for Administrator in the softphone? On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke [EMAIL PROTECTED] wrote: Go to the web page, in Preferences there are two pull down menus for Audio Input and Autio Output. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. Sent: Tuesday, February 08, 2005 2:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] snom soft phone Hi, How do I change the default audio device ? I have one of those USB headset (which actually is another soundcard) but the simulation insist in using my Soundblaster Live card :( -- Juanjo sin .sig :( ___ 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 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] why asterisk and ser
Good day all Why would u use asterisk and ser together and what is the big difference? 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] BRI only 2 calls
Good day all I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3 This is to install my quad bri card All installed well I coped over some old config files.All 4 ports are available,so that gives 8 open lines for incoming or outgoing,correct me of I'm rond The problem is,asterisk can only handle 2 calls at a time if there is 1 incoming(into pstn) and there someone already made a call out of the pstn,you cant make any other calls out or in On the cli it just show,when you try dialing out,Zap/4-1 got Hangup Even when you change the channels in zapata.conf,it keeps on showing,trying to make call Zap/10-1/012020121.Zap/10-1 got hangup? All the zttool and zttest shows its up and working Can this be a Telecoms provider problem please 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] asterisk remote monitor
Good day all We have a few remote pbx systems running I would like to monitor the and check that they are up and running and working 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] Dialplane slip
Good day all My extensions.conf is something like this [main] ;---incoming+ play welcome message extens = s.. ;---users extensions exten = 100. ;---outgoing ignore 0 ;- It all works fine The message says dial 1 for this ens But if I dial 0+number it will actually make a outgoing call! How do I stop this? I must allow the ignore 0 for internal uses but not if a call comes in from the outside? Please 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] h323
Good day all I have a asterisk server running sip and sip phone How do I get asterisk to call another h323 server? 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] h323 client
Good day all Just to re phrase my previous question We have asterisk running sip for sip phone In the US there is a h323 server What I want to do is: All calls coming into my pbx via sip thats got a american number to go threw the h323 server I have set this up with 2 sip servers where the one becomes a client? How do I do this with h323? 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] Grandstreams+Nat
Good day all I cant get my grandstream bt-100 to register My asterisk is on a public ip and the phone behind a nat firewall I added nat=yes in sip.conf and did this on my grandstream set the GS to SIP server=asterisk.yourhost.com and leave Outbound Proxy empty * set the GS to SIP port 5060 and RTP port 5004 (and Use random port=No) * set the GS to NAT traversal=Yes with STUN server=stun.xten.net * arrange port forwarding on your NAT router for tcp/udp 5062 and udp 5004 to your GS phone's IP address * enter nat=yes and canreinvite=no in your sip.conf for this GS phone user * in this order: issue a reload in the Asterisk console, restart your NAT router and reboot the phone * * 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] ilbc high bandwidth
Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well Is there a way to bring it down? Pleas 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] sip-sip
Good day all We have a asterisk server running sip for about 20 users We have a client running a unknown sip server in a different country I phone the guy there and he gave a a account(username+password) What I want is if a users calls the number of that country it should be send to the sip server on that side? What do i need to do on my side?Sip.conf and extensions.conf! Something like this? exten = ,1,Dial(SIP/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) is there something that I should do on sip.conf ore something? 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] Grandstream bt-100 loosing it!
Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145' 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] snom220
Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? 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] snom220
Sorry It works Just had to reboot the phone On Thu, 2005-01-13 at 08:40, Altus Snyman wrote: Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? 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 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] error?
Good day all I'm getting this error out of the blue on a incoming call? Any idea?Pleas Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ILBC since our native format has changed to SLINR ___ 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 e-mail spandsp
Did anyone get asterisk to actually work with a fax coming in on a pri number and e-mail it to a user? On Mon, 2005-01-10 at 08:29, Howard Lowndes wrote: On Mon, 2005-01-10 at 16:00, Altus Snyman wrote: Its still fails! [EMAIL PROTECTED] apps]# patch apps_makefile.patch.new patching file Makefile Hunk #1 succeeded at 42 with fuzz 2 (offset -7 lines). Hunk #2 FAILED at 73. 1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Yep, I've just had this one, and fixed it. cd asterisk/apps Go look at Makefile.rej and lines 19 20 (minus the leading + sign) are the ones that didn't make it into Makefile. If you put them in manually in the correct place then it all works. On Fri, 2005-01-07 at 22:08, Jim Radford wrote: Basically the changes in the apps/Makefile have progressed while the patch makefile have not. Here is a current patch that works as of CVS-HEAD-01/06/05-14:47:06 Regards, Jim On Fri, 7 Jan 2005, Altus Snyman wrote: I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on ___ 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] fax e-mail spandsp
Its still fails! [EMAIL PROTECTED] apps]# patch apps_makefile.patch.new patching file Makefile Hunk #1 succeeded at 42 with fuzz 2 (offset -7 lines). Hunk #2 FAILED at 73. 1 out of 2 hunks FAILED -- saving rejects to file Makefile.rej On Fri, 2005-01-07 at 22:08, Jim Radford wrote: Basically the changes in the apps/Makefile have progressed while the patch makefile have not. Here is a current patch that works as of CVS-HEAD-01/06/05-14:47:06 Regards, Jim On Fri, 7 Jan 2005, Altus Snyman wrote: I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on ___ 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] TE110P error
Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:9131 setup_zap: Unknown signalling method 'pri_cpe' Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:8789 setup_zap: Signalling must be specified before any channels are. Jan 10 08:17:18 WARNING[-1084595552]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Jan 10 08:17:18 WARNING[-1084595552]: loader.c:429 load_modules: Loading module chan_zap.so failed! 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] TE110P error
I'm getting this error now [chan_zap.so]Jan 10 09:05:05 WARNING[-1084362080]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_dump_info Jan 10 09:05:05 WARNING[-1084362080]: loader.c:429 load_modules: Loading module chan_zap.so failed! On Mon, 2005-01-10 at 08:42, Steven Critchfield wrote: On Mon, 2005-01-10 at 01:33 -0500, Alexander Lopez wrote: You are using a PRI based config for POTS lines. It will no worky. Post your zap*.conf files. I'll take a look at them for you.. How do you plug analog lines into a T1/E1 card? A better guess is either the driver for the card isn't loaded or the zap config files aren't agreeing with each other. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: Monday, January 10, 2005 1:24 AM To: asterisk Subject: [Asterisk-Users] TE110P error Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:9131 setup_zap: Unknown signalling method 'pri_cpe' Jan 10 08:17:18 ERROR[-1084595552]: chan_zap.c:8789 setup_zap: Signalling must be specified before any channels are. Jan 10 08:17:18 WARNING[-1084595552]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Jan 10 08:17:18 WARNING[-1084595552]: loader.c:429 load_modules: Loading module chan_zap.so failed! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax e-mail spandsp
I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax to email
Good day all I have a pri card,e100 What I want to do is If a fax comes in for number 1234567890 it should be e-mail to [EMAIL PROTECTED] If a fax comes in for number 0987654321 it should be e-mail to [EMAIL PROTECTED] ens Can this be done and how ___ 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 to email
and email-fax??
The other way around
On Thu, 2005-01-06 at 14:17, Andrew Kohlsmith wrote:
On January 6, 2005 06:55 am, Altus Snyman wrote:
Good day all
I have a pri card,e100
What I want to do is
If a fax comes in for number 1234567890 it should be e-mail to
[EMAIL PROTECTED]
If a fax comes in for number 0987654321 it should be e-mail to
[EMAIL PROTECTED]
ens
Yup it's easy. There are examples of how to effectively deal with faxes on
www.voip-info.org, and then you just combine the macro given there with some
extension magic like this:
exten = 1234567890,1,Macro(receive-fax,[EMAIL PROTECTED])
exten = 0987654321,1,Macro(receive-fax,[EMAIL PROTECTED])
etc.
Or you could have the receive-fax macro look up the email address from the
extension received... something like
exten = 1234567890,1,Macro(receive-fax,${EXTEN})
and then inside the macro, something like
exten = s,n,DBGet(EMAILTO,${ARG1})
...
exten = s,n,system(sendmail ${EMAILTO}, ${FAXFILENAME})
These are just pseudocode examples -- you need to look at the receive-fax
macro from www.voip-info.org or even
http://scottstuff.net/scott/archives/000152.html (found with google terms
receive fax asterisk), although the latter needs a little updating to work
with current Asterisk.
Also note that app_rxfax is *VERY* touchy about the version of the library
libtiff that is on your system. This is not an app_rxfax problem, libtiff
has some bugs when dealing with fax images in certain versions. Follow the
directions for building app_rxfax and spandsp very carefully, as they are
rather rigid.
I used to have segfault issues all the time with app_rxfax -- I have now
received well over a thousand faxes without a single crash.
-A.
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Re: [Asterisk-Users] fax to email
How do I fax a .tiff file with asterisk? On Thu, 2005-01-06 at 15:13, Michael Welter wrote: Altus Snyman wrote: and email-fax?? The other way around You can run a simple mail server on the * box to accept emails addressed to the .fax domain (i.e. [EMAIL PROTECTED]). This presumes you are able to forward the .fax domain from your main mail server to the * box. Once you have the email at the * box, it is a simple matter to convert the .ps or .pdf attachment to .tiff and send it to the fax machine. This method requires that the user convert the document to .ps or .pdf before attaching it to the email. The second method is print-to-fax. This requires the configuration of a Samba printer on the * box. Using the print function in MS Office (Word, Excel, etc.) client, the user would print a document to the printer. At the Samba interface, the .ps document would be captured, converted to .tiff, and sent. This method requires that the user embed the cover sheet, including the fax number into the document. The third method is to give your users a Windows printer plug-in that would send the cover sheet information along with the document (I'm working on this as we speak). This plug-in allows the user to send an address book entry along with the document, and the address book information is then used to compose the cover sheet. Cheers, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call(out) routing
Good day all
I had a look at the extensions.conf sorting
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting
What I'm trying to do is route all my cellphone number threw a channel
and all other calls threw the other 3 channels
Cellphone numbers are 10 number,i.o.w XX.
This is what I tried but it doesn't seem to work,please help
Thanks
Altus
extensions.conf
[cell_calls]
exten = _0083XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0083XXX,2,Congestion
exten = _0082XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0082XXX,2,Congestion
exten = _0084XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0084XXX,2,Congestion
exten = _0073XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0073XXX,2,Congestion
exten = _0072XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0072XXX,2,Congestion
exten = _0074XXX,1,Dial(vpb/1-4/${EXTEN:1})
exten = _0074XXX,2,Congestion
[MAIN]
exten = s.
ignorepat = 0
include = cell_calls
exten = _0.,1,Dial(vpb/1-2/${EXTEN:1})
exten = _0.,2,Dial(vpb/1-3/${EXTEN:1})
exten = _0.,3,Dial(vpb/1-1/${EXTEN:1})
exten = _0.,4,Congestion
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