Re: [asterisk-users] Version compatibility question...
Is there any version matching doc? since it was changed to Dahdi I don't really know which version works with which. On Sun, Dec 5, 2010 at 12:35 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Dec 02, 2010 at 09:09:25PM -0300, equis software wrote: Hi, Could I install Asterisk 1.4.19, Dahdi 2.4.0 and libpri 1.4.3 ?? No. Asterisk 1.4.22 cannot use DAHDI. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
I wouldn't bother with their hardware. You can run it on most servers providing the drivers for the hardware are supported. Just install it on a box with two NICs and put it between the router and your LAN, both static IPs, simple If I were you, I would find out what kind of DSL modem you have, but if it is doing NAT, DHCP, and all of that, you may be able to turn off everything except for the modem and use Vyatta for everything from NAT, DHCP, QoS, Squid, Firewall. In this case, one NIC would have your public IP, I suspect you would get it via DHCP or worst case, from your ISP, the second NIC is for the LAN, you can add more NICs for various purposes as well. I run Asterisk on Vyatta systems and it works great. No NAT issues with remote phones, QoS, and whatever else your imagination can come up with. I also install Webmin and NTOP. Just be aware that as soon as you activate the firewall, everything is blocked, so if you are going to use it as a firewall, get as many rules in place as you can think of. Thanks, Steve T On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five, except for solar flares, sandstorms, rain. Things beyond my control. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HA8 cards and RED alarm
Hi,
I have 2 servers: one is running 2 B410P cards with 8 euroisdn lines
(mISDN) connected on it, everything runs fine.
I prepare a new server - HP 360 G8- with 2 HA8 cards each of them 1
module of 4 lines. Already had with this machine an RMA on both cards as
they was faulty and crashed the server.
What happends is that when I connect cables on the HA8 modules (those
cables are unpluged from working server and connected to the new one)
nothing happend on the dahdi status, alarm is RED. Two days ago one
cable changed his staus to YELLOW (?) and then became again RED.
Below are relevant outputs. I created those config files with one of the
previous card which worked a short time and it was OK.
Could it be possible that modules have also to go for RMA?
Thanks for any hint.
SrvPhone2:/etc/asterisk# cat chan_dahdi.conf
;
; DAHDI telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the DAHDI channels
; CLI reload chan_dahdi.so
; will reload the configuration file,
; but not all configuration options are
; re-configured during a reload.
[channels]
;
; Default language
;
language=fr
;
; Default context
;
context=isdn
internationalprefix = 00
nationalprefix = 0
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
switchtype=euroisdn
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more
likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
;
callprogress=yes
progzone=be
; For fax detection, uncomment one of the following lines. The default
is *OFF*
; We use NVFaxDetect stuff for this
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
group=1
signalling=bri_cpe
context=isdn
channel = 1,2,4,5,7,8,10,11,13,14,16,17,19,20,22,23
SrvPhone2:/etc/dahdi# cat system.conf
loadzone = be
defaultzone = be
span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3
span = 2,2,0,ccs,ami
bchan = 4,5
hardhdlc = 6
span = 3,3,0,ccs,ami
bchan = 7,8
hardhdlc = 9
span = 4,4,0,ccs,ami
bchan = 10,11
hardhdlc = 12
span = 5,5,0,ccs,ami
bchan = 13,14
hardhdlc=15
span = 6,6,0,ccs,ami
bchan = 16,17
hardhdlc = 18
span = 7,7,0,ccs,ami
bchan = 19,20
hardhdlc = 21
span = 8,8,0,ccs,ami
bchan = 22,23
hardhdlc = 24
SrvPhone2*CLI dahdi show status
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
--
Daniel
--
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Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
Dear Steve; I am fully thanks for your advise and kindly help. I am asking about the ability to use vyatte hardware DSL router because of the following reasons: 1) I am afraid to make Asterisk the gateway for the whole network and this might effect on the performance and might cause a big load, u do not think so? 2) If any problem happened regarding to the QoS rules or regarding to the firewall or any other thing and they decided to do hardware restart for the server (or the PC machine), then the Asterisk will be restarted and that will effect on the telephony service at the site? 3) I am afraid if we applied the QoS and bandwidth divsion at Vyatte, and then we route the traffic to the DSL router (which will do the NAT to ISP), then all the QoS rules will be ignored (or become not effected)? What do u think? Again, special thanks for the guide and special help. Regards Bilal - I wouldn't bother with their hardware. You can run it on most servers providing the drivers for the hardware are supported. Just install it on a box with two NICs and put it between the router and your LAN, both static IPs, simple If I were you, I would find out what kind of DSL modem you have, but if it is doing NAT, DHCP, and all of that, you may be able to turn off everything except for the modem and use Vyatta for everything from NAT, DHCP, QoS, Squid, Firewall. In this case, one NIC would have your public IP, I suspect you would get it via DHCP or worst case, from your ISP, the second NIC is for the LAN, you can add more NICs for various purposes as well. I run Asterisk on Vyatta systems and it works great. No NAT issues with remote phones, QoS, and whatever else your imagination can come up with. I also install Webmin and NTOP. Just be aware that as soon as you activate the firewall, everything is blocked, so if you are going to use it as a firewall, get as many rules in place as you can think of. Thanks, Steve T On Thu, Dec 2, 2010 at 3:14 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I understood that Vyatta is the solution for the QoS, but I am not able to know if I can use a Vyatta hardware router to be DSL router and I set my QoS in it to resolve the voice problem. Is it possible? Thanks for the help. Regards Bilal Thanks all for ur participation and kindly advise. As I noticed that jitterbuffer could help if the ping does not have request time out but the voice is also cutting .. but in that case, I have to set the jitterbuffer at the IP Phones and Asterisk boxes. I have a polycom phone for example, and to set the jitterbuffer there are the following paramters: Payload Size Jitter Buffer Minimum Jitter Buffer Shrink Jitter Buffer Maximum When it use the minimum, and when it use the Shrink and when it use the maximum? If to look at the asterisk (in the SIP or IAX files) then there are a paramters for the jitterbuffer also, but really I am not able to know when to use this and when to use this: jenable, jbforce, jbmaxsize, jbresyncthreashold, jbimpl, jblog How to use the jbresyncthreashold? In which case? Regarding to the QoS, which will be need in case having a packet loose, correct? I just need to ask about something: What I will be able to do if my ISP did not setup the QoS at his side? What kind of settings I can do in my DSL router (in case of Cisco, or in case of Linksys that running linux firmware)? From the other side, if I used linux server to set the QoS, so do I have to let all the network elements to pass this linux server (so it will be the default gateway for other elements)? Appreciate the kindly help. Regards Bilal If getting a second circuit is out of the question. 1. Switch to SIP 2. Install and Learn Vyatta for QoS (Squid may help you quite a bit as well) as your router (or whatever you prefer) I use the paid versions of Vyatta but the free edition should be sufficient. I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping times. I used GSM and some tricks on the Vyatta box. Originally, before I deployed the above, it was a wild west situation like what you have now. Going from G729 to GSM made a big improvement in conjunction with QoS. My theory on that is that G729 is already a very lossy codec, so any more loss, garbled audio. GSM is less lossy. Switch from IAX to SIP was another huge improvement, and then finally putting Vyatta and QoS as my router made calls almost crystal clear. There was the obvious lag time but users get used to that and wait a second or two before speaking so they don't talk over each other and the quality was five by five,
Re: [asterisk-users] HA8 cards and RED alarm
2010/12/5 Administrator TOOTAI ad...@tootai.net
Hi,
I have 2 servers: one is running 2 B410P cards with 8 euroisdn lines
(mISDN) connected on it, everything runs fine.
I prepare a new server - HP 360 G8- with 2 HA8 cards each of them 1
module of 4 lines. Already had with this machine an RMA on both cards as
they was faulty and crashed the server.
What happends is that when I connect cables on the HA8 modules (those
cables are unpluged from working server and connected to the new one)
nothing happend on the dahdi status, alarm is RED. Two days ago one
cable changed his staus to YELLOW (?) and then became again RED.
Below are relevant outputs. I created those config files with one of the
previous card which worked a short time and it was OK.
Could it be possible that modules have also to go for RMA?
Thanks for any hint.
SrvPhone2:/etc/asterisk# cat chan_dahdi.conf
;
; DAHDI telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the DAHDI channels
; CLI reload chan_dahdi.so
; will reload the configuration file,
; but not all configuration options are
; re-configured during a reload.
[channels]
;
; Default language
;
language=fr
;
; Default context
;
context=isdn
internationalprefix = 00
nationalprefix = 0
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
switchtype=euroisdn
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more
likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
; This feature can also easily detect false hangups. The symptoms of this
is
; being disconnected in the middle of a call for no reason.
;
callprogress=yes
progzone=be
; For fax detection, uncomment one of the following lines. The default
is *OFF*
; We use NVFaxDetect stuff for this
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
group=1
signalling=bri_cpe
context=isdn
channel = 1,2,4,5,7,8,10,11,13,14,16,17,19,20,22,23
SrvPhone2:/etc/dahdi# cat system.conf
loadzone = be
defaultzone = be
span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3
span = 2,2,0,ccs,ami
bchan = 4,5
hardhdlc = 6
span = 3,3,0,ccs,ami
bchan = 7,8
hardhdlc = 9
span = 4,4,0,ccs,ami
bchan = 10,11
hardhdlc = 12
span = 5,5,0,ccs,ami
bchan = 13,14
hardhdlc=15
span = 6,6,0,ccs,ami
bchan = 16,17
hardhdlc = 18
span = 7,7,0,ccs,ami
bchan = 19,20
hardhdlc = 21
span = 8,8,0,ccs,ami
bchan = 22,23
hardhdlc = 24
SrvPhone2*CLI dahdi show status
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
--
Daniel
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Which Dahdi version ?
I had to use latest trunk to have mine working.
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[asterisk-users] no audio
Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include = stdexten exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() my sip.conf file [general] context=default allowoverlap=no bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a NAT ro ;register = xxx:y...@carrier.callwithus.com register = xxx:y...@sip.callwithus.com [callwithus] type=friend host=sip.callwithus.com username=xxx secret=yyy qualify=no insecure=invite -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no audio
On Sun, 5 Dec 2010, Thomas Perron wrote: Any reason why I don't get audio on the channel after it rings and the end user picks up. exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) Re-read 'core show application dial' Where is your prompt to option 'A' ? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no audio
Steve, thanks for your note negative. no joy. removed the line to make is very basic. see below. [globals] CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus ;[general] [default] include = stdexten exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/44) exten = s,n,Wait(2) exten = s,n,Hangup() ~ On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 5 Dec 2010, Thomas Perron wrote: Any reason why I don't get audio on the channel after it rings and the end user picks up. exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) Re-read 'core show application dial' Where is your prompt to option 'A' ? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no audio
Un-top-posting... On Sun, 5 Dec 2010, Thomas Perron wrote: Any reason why I don't get audio on the channel after it rings and the end user picks up. exten = s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks)) On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com wrote: Re-read 'core show application dial' Where is your prompt to option 'A' ? On Sun, 5 Dec 2010, Thomas Perron wrote: negative. no joy. removed the line to make is very basic. see below. exten = s,1,Answer() exten = s,n,Wait(1) exten = s,n,Dial(SIP/callwithus/44) Crank up the verbosity and debugging levels, check the codecs, etc. Does 'sip set debug on' give any clues? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM calls fall after some minutes
Late reply: On Wed, Nov 24, 2010 at 12:19:36PM +0100, luca capra wrote: Hello, it's my first post on this list, I hope not to bore youwith my novice questions.. We're using a TDM400 with 3 fxo modules connected to pstn. Call goes inbound/outbound correctly, after playing a bit on some dahdi-channels.conf/chan_dahdi.conf options. The big problem is that after 5 minutes and 16'' of outbound call, suddenly the line fall down. Likely someone an hangup. I'm looking the signalling method (actually fxs_ks) but with no luck. Does somebody had similar problems ? Who decides to hang up? Any relevant console log messages? If not: enable debug (and debug logging) and see if that gives you some clues. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
