Re: [asterisk-users] spandsp (foip)
linksys spa3102 or 2100 are known to work. Grandstream also should do it with recent firmware. Don't be fooled by what is written on the box, lot of ata's out there claim t.38. (while the firmware doesnt contain anything related to t.38) Zoa Christopher Corn wrote: lee, Thanks for the feedback. in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case, where does the packetization of the t30 data happen? at the ATA? level i presume? http://www.answers.com/topic/t-30-protocol-figure-01-jpg also, can you recommened a good asterisk compatible ATA adapter with t38 support? i believe cisco has one. Thanks in advance. */Lee Howard <[EMAIL PROTECTED]>/* wrote: Christopher Corn wrote: > May I ask, from your own personal experience. is it not necessaritly > worth (the headaches) of investing mytime into setting up SPANDSP into > my asterisk system, but rather invest it into going to a company, like > packet8 that offers t38 conversion? I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing). If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.) > what does the future of faxing lean towards? before entering an era > when all fax machines run the t38 protocol. will we see more t38 > termination services or faxing through g711? T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon. > from what i've read, using a service that does t38 termination, seems > to be where i should go. I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you. Lee. ___ --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 ___ --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] Display message on voip phone...hint?
Hi, use AOC. See here: http://www.snom.com/wiki/index.php/FAQs#Q:_How_to_show_billing_information_on_the_phone_display.3F Regards, Sven On Friday 22 September 2006 17:31, Ale wrote: > Hi all, > > Can anyone help me... i need to display the "cost" of a call during a > conversation on a sip or iax phone. > > I see on voip-info that some snom phone support sendtext application, > but i don't know how to update the message with the cost on the phone > during the conversation. > > Every suggestion is apreciated. > > Thx, > Bye Bye Ale > ___ > --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 -- -- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 PSTN/ENUM +49 30 39833434 mailto:[EMAIL PROTECTED] http://www.snom.com -- -- --- See our Docs, FAQs, etc at: http://snom.com/wiki --- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.com --- ___ --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] spandsp (foip)
lee, Thanks for the feedback. in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case, where does the packetization of the t30 data happen? at the ATA? level i presume? http://www.answers.com/topic/t-30-protocol-figure-01-jpg also, can you recommened a good asterisk compatible ATA adapter with t38 support? i believe cisco has one. Thanks in advance. Lee Howard <[EMAIL PROTECTED]> wrote: Christopher Corn wrote:> May I ask, from your own personal experience. is it not necessaritly > worth (the headaches) of investing mytime into setting up SPANDSP into > my asterisk system, but rather invest it into going to a company, like > packet8 that offers t38 conversion?I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing).If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.)> what does the future of faxing lean towards? before entering an era > when all fax machines run the t38 protocol. will we see more t38 > termination services or faxing through g711?T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon.> from what i've read, using a service that does t38 termination, seems > to be where i should go.I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you.Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] High utilization with SIP registration
Greetings all, I have a problem with a PBX that I manage. The system has 2 AVM Fritz boards connected to two BRI ISDN services using chan_capi in addition to several SIP trunks going out to Internet based providers for call termination via the Internet. They experience problems when the Internet connection goes down. Obviously the SIP trunks are lost. However the strange thing is that calls are dropped on the capi channels as well during these Internet outages. One of the engineers that I work with felt that the problem was due to Asterisk persistantly trying to re register the SIP services and was using up too much CPU in the process. In fact he was able to workaround the problem temporarily by commenting out the SIP registration in sip.conf, which would confirm his theory. I suppose my question is. Has anyone else seen this sort of behaviour before? Is there any SIP settings that we should be including to try to slow down the SIP registration so that it doesn't use up too many system resources? This message was sent using MyMail ___ --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] iaxy: one way audio
Responding to my post for searching purposes; The fix is to manually specify disallow=all, allow=ulaw for each device. It does not seem to work if you only include that in the globals. Sean Sean Kennedy wrote: Hey all, So I just got an iaxy to play with a few days ago. Got the config files figured out and configured the device. I was able to make phone calls out on it just fine. However, when trying to call the device I get a one way audio problem ( which I would expect from sip, but not iaxy ). The user on the iaxy can hear but their audio isn't transmitted. I have double checked the iaxyprov config file, turning on heartbeat ( in case it's a firewall timeout problem ). I checked asterisk's iaxy.conf file, and all the ip information in there looks correct. I'm not sure how to procede to troubleshoot this problem. Any help is greatly appreciated. Sean iax260.conf: [EMAIL PROTECTED] trunk]# vi iax260.conf ; ; IAXY Provisioning description ; dhcp ;ip: 192.168.3.90 ;netmask: 255.255.255.0 ;gateway: 192.168.3.1 codec: ulaw ;codec: adpcm server: 192.168.1.7 ;altserver: 192.168.0.2 user: user pass: userpass register heartbeat ;debug ; ; Feature tuning (default is all enabled) ; ;disablecid ;disablecw ;disablecidcw ;disable3way iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 192.168.1.7; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [user] username=user type=friend secret=userpass record_out=Adhoc record_in=Adhoc qualify=no port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device trunk=no ___ --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] Missing sound in spanish from 1.4 beta2
- Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> I will assume that you are a native speaker; I'm not equipped to
> evaluate whether ... well, anyway. Anyone know where those prompts
> actually *came* from? :-)
>
> Cheers,
> -- jra
The Spanish language core-sounds came from Allison Smith. She is the same
person who does the English language sounds (You can get English and Spanish
language prompts from Allison Smith, or French language prompts from June
Wallack {who does the French language core-sounds prompts}, via
http://www.digium.com/en/products/voice/ - they both do very good work).
--
Jason Parker
Digium
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Re: [asterisk-users] Rpath PoundKey 1.2
Is there a way to correct the problem or can the files be generated? Did you run the registration program? Asterisk won't start unless it's registered with Digium. -Brian ___ --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] spandsp (foip)
Christopher Corn wrote: May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion? I am not really in a position to tell you what something will be worth to you - especially when I've not even used that something myself. I know and use spandsp as a library, with IAXmodem and HylaFAX, but I do not have any experience with spandsp in txfax/rxfax applications or in its new T.38 gatewaying. I suspect that I'll eventually get into spandsp's T.38 aspects, but without that I've only had a limited amount of hands-on exposure to T.38 applications in the form of t38modem and Cisco gateways (which experience was somewhat disenchanting - mostly because of the gateway T.30 processing). If you have a T.38 fax machine or if you have a T.38-capable ATA connected to a fax machine and you do not have your own PSTN lines then I would suspect that it would be worthwhile to use T.38 pass-through on Asterisk 1.4 or OpenPBX in conjunction with a T.38-supporting FoIP provider. (Because otherwise you don't have any straight-forward, reliable means for faxing from your internal fax machines.) what does the future of faxing lean towards? before entering an era when all fax machines run the t38 protocol. will we see more t38 termination services or faxing through g711? T.38 is the end-all solution for faxing over IP networks. So I suspect that you will see the pervasiveness of T.38 implementations increase along with the pervasiveness of VoIP in general. That said, VoIP has its own fair share of problems that keeps it from being capable of replacing PSTN circuits entirely, and so as long as those problems are not generally resolvable for your average business or service provider then you'll continue to also see more of the same, traditional, modem-ing fax machines. So I strongly suspect that you'll see more of T.38, but I don't think that the PSTN (and traditional fax machines with it) is going away any time soon. from what i've read, using a service that does t38 termination, seems to be where i should go. I would say that it entirely depends upon whether or not you have PSTN lines yourself. If you do, then I would take whatever efforts you can to avoid the additional points of T.30 processing/relaying (therefore avoiding T.38 gatewaying). But if you do not have PSTN lines, then take whatever efforts you can to properly implement T.38 to your FoIP provider who will gateway for you. Lee. ___ --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] Need a recommended T38 FOIP solution
My Asterisk PBX now is strictly IP, out a 6000/600kbps using the g729 codec. Only thing using pstn now are my fax machines. Thanks for the input Jay. "Jay R. Ashworth" <[EMAIL PROTECTED]> wrote: On Sun, Sep 24, 2006 at 08:06:41PM -0700, Christopher Corn wrote:> I help support a small office, 5 SIP phones, connected to an> Asterisk PBX. We have 4 analouge fax machines connected to a pstn> that i would like to get rid of, but need a foip solution.>> rather thing trying to do a pass through using the g711 protocol,> I want to go with a t38 termination since it is more reliable. can> someone recommened a cheap t38 foip vendor? also, what kind of> changes do i need to make to my analouge fax machines so that i can> get this accomplished? i assume theres some type of ATA adapter> that will need to be used with the phone. specific brand?>> To receive faxes I assume I could use asterfax,but to send faxes i> need to use a fax machine, mainly because people here will need to> sign documents then fax them.What is your PSTN uplink now, and what's it likely to be? I assumeyour primary uplink is still analog PSTN? Cause if it is, you *might*be able to use G.711 passthrough, since you won't be trying to send theG.711 over a VoIP link. On a LAN, it might work out for you.Lee may have a more informed opinion on this; I will admit to speculating.Cheers,-- jra-- Jay R. Ashworth [EMAIL PROTECTED]Designer Baylink RFC 2100Ashworth & Associates The Things I Think '87 e24St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274"That's women for you; you divorce them, and 10 years later,they stop having sex with you." -- Jennifer Crusie; _Fast_Women--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
Re: [asterisk-users] Missing sound in spanish from 1.4 beta2
On Sun, Sep 24, 2006 at 10:08:51PM -0500, Carlos Chavez wrote: > On Sun, 24 Sep 2006 16:47:09 -0400, Jay R. Ashworth wrote > > On Sun, Sep 24, 2006 at 12:33:01PM -0500, Carlos Chavez wrote: > > > I just installed 1.4 beta 2 with the spanish sound set. Apart from the > > > voice sounding definitively as a non native speaker > > > > "A speaker not native" to where? > > > > How many countries is Spanish the primary language in? > > > Not native to any spanish speaking country. It was recorded by someone > who Spanish is not their primary language and does not have a good > pronunciation. But hey, its free so can´t complain much. The point is that > they need to add vm-youhaveno.gsm so you can use the voicemail application. I will assume that you are a native speaker; I'm not equipped to evaluate whether ... well, anyway. Anyone know where those prompts actually *came* from? :-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] Need a recommended T38 FOIP solution
On Sun, Sep 24, 2006 at 08:06:41PM -0700, Christopher Corn wrote: >I help support a small office, 5 SIP phones, connected to an >Asterisk PBX. We have 4 analouge fax machines connected to a pstn >that i would like to get rid of, but need a foip solution. > >rather thing trying to do a pass through using the g711 protocol, >I want to go with a t38 termination since it is more reliable. can >someone recommened a cheap t38 foip vendor? also, what kind of >changes do i need to make to my analouge fax machines so that i can >get this accomplished? i assume theres some type of ATA adapter >that will need to be used with the phone. specific brand? > >To receive faxes I assume I could use asterfax,but to send faxes i >need to use a fax machine, mainly because people here will need to >sign documents then fax them. What is your PSTN uplink now, and what's it likely to be? I assume your primary uplink is still analog PSTN? Cause if it is, you *might* be able to use G.711 passthrough, since you won't be trying to send the G.711 over a VoIP link. On a LAN, it might work out for you. Lee may have a more informed opinion on this; I will admit to speculating. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] Missing sound in spanish from 1.4 beta2
On Sun, 24 Sep 2006 16:47:09 -0400, Jay R. Ashworth wrote > On Sun, Sep 24, 2006 at 12:33:01PM -0500, Carlos Chavez wrote: > > I just installed 1.4 beta 2 with the spanish sound set. Apart from the > > voice sounding definitively as a non native speaker > > "A speaker not native" to where? > > How many countries is Spanish the primary language in? > Not native to any spanish speaking country. It was recorded by someone who Spanish is not their primary language and does not have a good pronunciation. But hey, its free so can´t complain much. The point is that they need to add vm-youhaveno.gsm so you can use the voicemail application. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Need a recommended T38 FOIP solution
I help support a small office, 5 SIP phones, connected to an Asterisk PBX. We have 4 analouge fax machines connected to a pstn that i would like to get rid of, but need a foip solution. rather thing trying to do a pass through using the g711 protocol, I want to go with a t38 termination since it is more reliable. can someone recommened a cheap t38 foip vendor? also, what kind of changes do i need to make to my analouge fax machines so that i can get this accomplished? i assume theres some type of ATA adapter that will need to be used with the phone. specific brand? To receive faxes I assume I could use asterfax,but to send faxes i need to use a fax machine, mainly because people here will need to sign documents then fax them. any recommendation would be appreciated. thanks.___ --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] Re: 2 CPU's, Only 1 taking IRQ's
On Mon September 25 2006 11:05, Bart Fisher <[EMAIL PROTECTED]> wrote: > Hmm, this must not be installed: > # locate irqbalance > # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h > > How do I install this? > > Bart I'd run `apt-get install irqbalance`, but you'd do something with yum or whatever new-fangled thing CentOS uses. As always, Google is your friend, Bart. Please search before asking obvious questions. Cheers, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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] Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25
A better solution would be to modify the line mailhub=mail in /etc/ssmtp/ssmtp.conf such that mailhub points to your smtp server. -- Bernhard > Hey, that's why i had no idea how to spot the glitch... I added a line in > my /etc/hosts file for "mail" aimed at my SMTP server, all better now. > > Thanks! > > Nick > > > -- > Nick Ellson > CCDA, CCNP, CCSP, CCAI, > MCSE 2000, Security+, Network+ > Network Hobbyist, VFR Private Pilot. > > > On Sun, 10 Sep 2006, C F wrote: > >> Take this to sendmail list. this is not an asterisk problem. In any >> case it looks like it's trying to send email to host mail on port 25 >> and it's failing. Try doing a telnet mail 25 and see what happens. >> >> On 9/10/06, Nick Ellson wrote: >>> >>> OK, help.. Am not sure where this is not configured right. I followed the >>> voicemail.conf directions, even tried specifying sendmail -t directly. >>> >>> My sendmail mail log shows: >>> >>> Sep 10 17:19:26 goonie sSMTP[4439]: Unable to locate mail >>> Sep 10 17:19:26 goonie sSMTP[4439]: Cannot open mail:25 >>> >>> Nick >>> >>> >>> >>> -- >>> Nick Ellson >>> CCDA, CCNP, CCSP, CCAI, >>> MCSE 2000, Security+, Network+ >>> Network Hobbyist, VFR Private Pilot. >>> >>> ___ >>> --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 ___ --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] spandsp (foip)
Lee and everyone else that replied, Thanks for the valuable and very detailed explanation. May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion? what does the future of faxing lean towards? before entering an era when all fax machines run the t38 protocol. will we see more t38 termination services or faxing through g711? from what i've read, using a service that does t38 termination, seems to be where i should go. Thanks. Lee Howard <[EMAIL PROTECTED]> wrote: On Sun, Sep 24, 2006 at 01:58:21PM -0700, Christopher Corn wrote:> A couple of faxing methods im confused about.>> The pass through method, sending fax data over G711 codec> versus> Relay method, t30 to t38 conversion>>> Can someone explain to me why the pass through method doesn't> require t30 to t38 conversion ( or does it do it?)? i believe> the conversion to t38 is so that it can be routed through a> packet network and then back to t30 so that the fax machine can> understand. why is it that if you use a pass through method, and> your still passing through a packet network, you dont need to> convert to t38 and t30?>Be careful about your wording. People here generally refer to "pass through" as T.38 pass-through and not G.711 pass-through.I think that if you understood how faxing works you would see that your questions here don't really make sense.In traditional PSTN faxing you have a total of two endpoints performing T.30 protocol. In a simplified form, the sender takes scanner image data and modulates it (into an audio waveform) and then passes that audio over the PSTN to the receiver which demodulates it (takes the audio and turns it into data again). As long as the demodulated data is identical to the original data, then everything should be okay... for the most part. However, if you start to consider audio corruption on the PSTN, then that's where difficulties start to ensue. If you have some audio, modulated data, and then you compress it or fracture it or otherwise corrupt it, then there's no possible way that the demodulator is going to be able to come up with the original data.Now introduce VoIP telephony... where a small amount of audio corruption (jitter) is anticipated on the UDP channel... and mix it with faxing and hopefully you can see how it just doesn't work well. VoIP is packetized audio passed over an IP network. Packetized audio is nothing new. ISDN circuits have had it for a long time now. Those circuits are digital - meaning the audio waveform is digitized at 8000 Hz... so the audio is represented with bytes and are packetized into frames. Those traditional digital circuits are designed to prevent any loss of that data. VoIP works similarly, except that the medium is lossy UDP/IP networking.Since VoIP works on *IP* networks, and since IP networks already handle data communication very well, there really is no reason to perform the modulation or the demodulation - just send the raw data through. So that's basically the punchline of T.38... it's fax protocol without the traditional modems involved. Then you have FoIP.However, these days the world is a hybrid of VoIP and PSTN environments (mostly PSTN still), and thus anyone using T.38 will need to have a "gateway" involved somewhere along the call path that can do that traditional modulation/demodulation. That is what the T.38 gateway is. If a T.38 relay does not act as a gateway (i.e. no modulators) then it performs only T.38 pass-through - meaning it only is useful for situations where calls are end-to-end T.38 or where an external FoIP service provider is used.Because of the way things work T.38 gateways will not only need to have traditional modems (hard or soft) but will also need to perform T.30. So when faxing with T.38 and the call is not end-to-end T.38 then you have at least three points along the call path performing T.30 (versus the traditional scenario of just two).So, to answer your questions...Why does using G.711 not require T.38? Because from the viewpoint that the question was given, G.711 and T.38 are competing approaches. T.38 was designed to replace G.711. You can packetize G.711 audio just fine without converting it to anything else. So when faxing with G.711 T.38 is not involved because its basically mimicking the old-style traditional PSTN faxing, except that the audio is passing over a different (less-reliable) medium.So the reason that T.38 exists is because UDP/IP is lossy and is not therefore reliable for the purposes of faxing with G.711 unless the communication can be guaranteed to be nearly lossless. For those that work on lossy channels, G.711 will just not work reliably.Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options
Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. ___ --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] e911
I'm keeping my Qwest line for this purpose.On 9/23/06, Christopher Corn <[EMAIL PROTECTED]> wrote: Im using voipestreet and voxee for my SIP termination. neither of them, offer any kind of e911 service. as i search the web i see different companies that offer this e911 service to voip suppliers. I want to choose the right one, seeing how in an emergency, it can be very crucial. any suggestions? thanks. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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
Re: [asterisk-users] spandsp (foip)
On Sun, Sep 24, 2006 at 05:17:22PM -0700, Lee Howard wrote: > So, to answer your questions... > > Why does using G.711 not require T.38? Because from the viewpoint that > the question was given, G.711 and T.38 are competing approaches. T.38 > was designed to replace G.711. You can packetize G.711 audio just fine > without converting it to anything else. So when faxing with G.711 T.38 > is not involved because its basically mimicking the old-style > traditional PSTN faxing, except that the audio is passing over a > different (less-reliable) medium. > > So the reason that T.38 exists is because UDP/IP is lossy and is not > therefore reliable for the purposes of faxing with G.711 unless the > communication can be guaranteed to be nearly lossless. For those that > work on lossy channels, G.711 will just not work reliably. Lee's answer is more complete than mine (as you might expect from the fact that he does this stuff for money, and I only speculate about it :-). I wasn't thinking about jitter and packet loss... though clearly I should have, since Fax is a WAN app. (I'm embroiled in a design project for a big Asterisk switch, and I'm thinking "over the LAN" this week. Sorry. :-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] spandsp (foip)
On Sun, Sep 24, 2006 at 01:58:21PM -0700, Christopher Corn wrote: A couple of faxing methods im confused about. The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversion Can someone explain to me why the pass through method doesn't require t30 to t38 conversion ( or does it do it?)? i believe the conversion to t38 is so that it can be routed through a packet network and then back to t30 so that the fax machine can understand. why is it that if you use a pass through method, and your still passing through a packet network, you dont need to convert to t38 and t30? Be careful about your wording. People here generally refer to "pass through" as T.38 pass-through and not G.711 pass-through. I think that if you understood how faxing works you would see that your questions here don't really make sense. In traditional PSTN faxing you have a total of two endpoints performing T.30 protocol. In a simplified form, the sender takes scanner image data and modulates it (into an audio waveform) and then passes that audio over the PSTN to the receiver which demodulates it (takes the audio and turns it into data again). As long as the demodulated data is identical to the original data, then everything should be okay... for the most part. However, if you start to consider audio corruption on the PSTN, then that's where difficulties start to ensue. If you have some audio, modulated data, and then you compress it or fracture it or otherwise corrupt it, then there's no possible way that the demodulator is going to be able to come up with the original data. Now introduce VoIP telephony... where a small amount of audio corruption (jitter) is anticipated on the UDP channel... and mix it with faxing and hopefully you can see how it just doesn't work well. VoIP is packetized audio passed over an IP network. Packetized audio is nothing new. ISDN circuits have had it for a long time now. Those circuits are digital - meaning the audio waveform is digitized at 8000 Hz... so the audio is represented with bytes and are packetized into frames. Those traditional digital circuits are designed to prevent any loss of that data. VoIP works similarly, except that the medium is lossy UDP/IP networking. Since VoIP works on *IP* networks, and since IP networks already handle data communication very well, there really is no reason to perform the modulation or the demodulation - just send the raw data through. So that's basically the punchline of T.38... it's fax protocol without the traditional modems involved. Then you have FoIP. However, these days the world is a hybrid of VoIP and PSTN environments (mostly PSTN still), and thus anyone using T.38 will need to have a "gateway" involved somewhere along the call path that can do that traditional modulation/demodulation. That is what the T.38 gateway is. If a T.38 relay does not act as a gateway (i.e. no modulators) then it performs only T.38 pass-through - meaning it only is useful for situations where calls are end-to-end T.38 or where an external FoIP service provider is used. Because of the way things work T.38 gateways will not only need to have traditional modems (hard or soft) but will also need to perform T.30. So when faxing with T.38 and the call is not end-to-end T.38 then you have at least three points along the call path performing T.30 (versus the traditional scenario of just two). So, to answer your questions... Why does using G.711 not require T.38? Because from the viewpoint that the question was given, G.711 and T.38 are competing approaches. T.38 was designed to replace G.711. You can packetize G.711 audio just fine without converting it to anything else. So when faxing with G.711 T.38 is not involved because its basically mimicking the old-style traditional PSTN faxing, except that the audio is passing over a different (less-reliable) medium. So the reason that T.38 exists is because UDP/IP is lossy and is not therefore reliable for the purposes of faxing with G.711 unless the communication can be guaranteed to be nearly lossless. For those that work on lossy channels, G.711 will just not work reliably. Lee. ___ --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: [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem
On Sun, Sep 24, 2006 at 02:01:01PM -0700, Lee Howard wrote: > Artifex Maximus wrote: > >zttest is often on 99.975586% with final result: > >--- Results after 67 passes --- > >Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764 > > This is unacceptable for faxing, and it is evidence of the underlying > problem also causing your faxes to come through with poor quality. > > > 0: 2087872259IO-APIC-edge timer > > 7: 0IO-APIC-edge parport0 > > 8: 1IO-APIC-edge rtc > > 9: 1 IO-APIC-level acpi > >14: 18440124IO-APIC-edge ide0 > >15:4456445IO-APIC-edge libata > >169:4878102 IO-APIC-level eth0 > >177: 2086847525 IO-APIC-level wctdm24xxp > >185: 2086810653 IO-APIC-level wct4xxp > > Notice the priorities here... and that your Zaptel cards come *last*, > after eth0, after IDE. Each of those Zap cards are going to generate an > interrupt once every millisecond when in use. You can hopefully imagine > how IDE or eth0 activity would interfere, since they have a higher > priority than the Zap cards. The Digium cards interrupt *on every scheduler tick*? Is that a latency thing? or just sloppy design? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] spandsp (foip)
Christopher Corn wrote: I've been reading about FOIP and there something i dont undrestand. maybe someone can explain to me. A couple of faxing methods im confused about. The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversion Can someone explain to me why the pass through method doesn't require t30 to t38 conversion ( or does it do it?)? i believe the conversion to t38 is so that it can be routed through a packet network and then back to t30 so that the fax machine can understand. why is it that if you use a pass through method, and your still passing through a packet network, you dont need to convert to t38 and t30? hope this makes sense. thanks alot. See http://www.soft-switch.org/foip.html Steve ___ --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] spandsp (foip)
On Sun, Sep 24, 2006 at 01:58:21PM -0700, Christopher Corn wrote: >I've been reading about FOIP and there something i dont undrestand. maybe >someone can explain to me. > >A couple of faxing methods im confused about. > >The pass through method, sending fax data over G711 codec >versus >Relay method, t30 to t38 conversion > > >Can someone explain to me why the pass through method doesn't >require t30 to t38 conversion ( or does it do it?)? i believe >the conversion to t38 is so that it can be routed through a >packet network and then back to t30 so that the fax machine can >understand. why is it that if you use a pass through method, and >your still passing through a packet network, you dont need to >convert to t38 and t30? I believe the proper answer to your question is this: Fax and modem tones will not be passed properly by some codecs because they're optimized for the frequency patterns of human voices -- and because those tones were themselves optimized for a traditional analog or DS-0 PCM channel. Since G.711 *is* an uncompressed DS-0, that's why you can pass fax tones through it, without any help from spoofing protocols like t.38. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] dialout-trunk vs. dial group
On Fri, Sep 22, 2006 at 09:16:45AM -0600, Nathan Bell wrote:
> Hi everybody,
>
> Is there any significant difference between using
> Macro(dialout-trunk,1,${EXTEN}) and Dial(Zap/g1/${EXTEN})? If so, what
> are the differences?
>
> I am not using freePBX, or any variant of it, but want the
> functionallity of dialout-trunk. If I define the trunk in zapata.conf,
> will using Dial() suffice?
Macro(dialout-trunk,1,${EXTEN}) basically jumps into the context
macro-dialout-trunk (but also assigns some variables, and returns
later). So it really depends on what that specific macro does.
What extra functionality do you need?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
___
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Re: [asterisk-users] Trixbox Documentation
joea, j4computers wrote: > So, now I am struggling with a Suse SLES 9 install, that seems reluctant to > co-operate. I have a number of boxes running CentOS 4.4 with Asterisk 1.2 and FreePBX: Because I install everything manually, I know it all works, without the overhead of the Trixbox features I have no intention of ever using (a2billing, Sugar, etc). I find that following the FreePBX install procedures for CentOS to be quite straight-forward. I have a bunch of Digium and Sangoma cards as well, all working too. My 2c, YMMV, etc. Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --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] Rpath PoundKey 1.2
Is the Pound Key 1.2 Asterisk broken on install. I installed ver. 1.2 and every time I try to run Asterisk -r, I receive a cannot connect to remote Asterisk, does /var/run/asterisk.ctl exist. Is there a way to correct the problem or can the files be generated? Thanks, Bobby ___ --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: [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Artifex Maximus wrote: zttest is often on 99.975586% with final result: --- Results after 67 passes --- Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764 This is unacceptable for faxing, and it is evidence of the underlying problem also causing your faxes to come through with poor quality. 0: 2087872259IO-APIC-edge timer 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 18440124IO-APIC-edge ide0 15:4456445IO-APIC-edge libata 169:4878102 IO-APIC-level eth0 177: 2086847525 IO-APIC-level wctdm24xxp 185: 2086810653 IO-APIC-level wct4xxp Notice the priorities here... and that your Zaptel cards come *last*, after eth0, after IDE. Each of those Zap cards are going to generate an interrupt once every millisecond when in use. You can hopefully imagine how IDE or eth0 activity would interfere, since they have a higher priority than the Zap cards. Lee. ___ --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] spandsp (foip)
I've been reading about FOIP and there something i dont undrestand. maybe someone can explain to me. A couple of faxing methods im confused about. The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversion Can someone explain to me why the pass through method doesn't require t30 to t38 conversion ( or does it do it?)? i believe the conversion to t38 is so that it can be routed through a packet network and then back to t30 so that the fax machine can understand. why is it that if you use a pass through method, and your still passing through a packet network, you dont need to convert to t38 and t30? hope this makes sense. thanks alot.___ --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] Missing sound in spanish from 1.4 beta2
On Sun, Sep 24, 2006 at 12:33:01PM -0500, Carlos Chavez wrote: > I just installed 1.4 beta 2 with the spanish sound set. Apart from the > voice sounding definitively as a non native speaker "A speaker not native" to where? How many countries is Spanish the primary language in? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
>> It appears that CPU1 in not taking any interrupts - What steps do I >> need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. -- Atte. Álvaro Palma ___ --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] 2 CPU's, Only 1 taking IRQ's
Sunday, September 24, 2006, 8:20:19 PM, Steve Totaro wrote: > Bart Fisher wrote: >> I'm trying to solve a echo problem... >> >> The system is Centos 2.6.9-34.0.2.ELsmp (SMP) CentOS release 4.3 >> (Final). And the Box is Dual Intel Xeon CPU 2.80GHz with 2 GB memory. >> >> It appears that CPU1 in not taking any interrupts - What steps do I >> need to do >> bring up CPU1 and share IRQ requests for a Linux noob? > Might be a dumb question but are you sure there are two procs? I got > confused once because of hyperthreading and looking at top. No. Linux kernel must configured irq balancing if the computer has more than once cpu, and you want to the other cpus grab irq-s too. -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --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 to configure a sip service
hi there. I am trying to configure a sip service on my asterisk that would answer and sent people to the [multi.start] part of my dialplan the sip Url I'd like this to answer to is [EMAIL PROTECTED] how would I configure that ? ___ --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] 2 CPU's, Only 1 taking IRQ's
Bart Fisher wrote: I'm trying to solve a echo problem... The system is Centos 2.6.9-34.0.2.ELsmp (SMP) CentOS release 4.3 (Final). And the Box is Dual Intel Xeon CPU 2.80GHz with 2 GB memory. It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Bart cat /proc/interrupts CPU0 CPU1 0: 70759112 0IO-APIC-edge timer 1:137 0IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 12: 2564 0IO-APIC-edge i8042 14: 173957 0IO-APIC-edge ide0 15: 635817 0IO-APIC-edge ide1 153: 0 0 IO-APIC-level uhci_hcd 161:1437910 0 IO-APIC-level eth0 169: 0 0 IO-APIC-level uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd 185: 70717012 0 IO-APIC-level wct4xxp 193: 70716921 0 IO-APIC-level wct4xxp NMI: 0 0 LOC: 70763311 70763320 ERR: 0 MIS: 0 Distro Name Might be a dumb question but are you sure there are two procs? I got confused once because of hyperthreading and looking at top. Thanks, Steve ___ --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] 2 CPU's, Only 1 taking IRQ's
I'm trying to solve a echo problem... The system is Centos 2.6.9-34.0.2.ELsmp (SMP) CentOS release 4.3 (Final). And the Box is Dual Intel Xeon CPU 2.80GHz with 2 GB memory. It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Bart cat /proc/interrupts CPU0 CPU1 0: 70759112 0IO-APIC-edge timer 1:137 0IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 12: 2564 0IO-APIC-edge i8042 14: 173957 0IO-APIC-edge ide0 15: 635817 0IO-APIC-edge ide1 153: 0 0 IO-APIC-level uhci_hcd 161:1437910 0 IO-APIC-level eth0 169: 0 0 IO-APIC-level uhci_hcd 177: 0 0 IO-APIC-level uhci_hcd 185: 70717012 0 IO-APIC-level wct4xxp 193: 70716921 0 IO-APIC-level wct4xxp NMI: 0 0 LOC: 70763311 70763320 ERR: 0 MIS: 0 Distro Name ___ --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] Missing sound in spanish from 1.4 beta2
I just installed 1.4 beta 2 with the spanish sound set. Apart from the voice sounding definitively as a non native speaker (heavy accent) there is a sound file missing from the voicemail set. The vm-youhaveno.gsm file is not included so you cannot get into the voicemail application if you have no mail waiting. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] iaxy: one way audio
Hey all, So I just got an iaxy to play with a few days ago. Got the config files figured out and configured the device. I was able to make phone calls out on it just fine. However, when trying to call the device I get a one way audio problem ( which I would expect from sip, but not iaxy ). The user on the iaxy can hear but their audio isn't transmitted. I have double checked the iaxyprov config file, turning on heartbeat ( in case it's a firewall timeout problem ). I checked asterisk's iaxy.conf file, and all the ip information in there looks correct. I'm not sure how to procede to troubleshoot this problem. Any help is greatly appreciated. Sean iax260.conf: [EMAIL PROTECTED] trunk]# vi iax260.conf ; ; IAXY Provisioning description ; dhcp ;ip: 192.168.3.90 ;netmask: 255.255.255.0 ;gateway: 192.168.3.1 codec: ulaw ;codec: adpcm server: 192.168.1.7 ;altserver: 192.168.0.2 user: user pass: userpass register heartbeat ;debug ; ; Feature tuning (default is all enabled) ; ;disablecid ;disablecw ;disablecidcw ;disable3way iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 192.168.1.7; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [user] username=user type=friend secret=userpass record_out=Adhoc record_in=Adhoc qualify=no port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device trunk=no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: 1.4 Beta 2 Config Problem
Thanks for the feedback. It is
working now. For those interested here is what I have that works
with 1.4 beta2
exten =>
8600,1,GotoIf($[${CALLERID(num)} = 2001]?50:2)
exten =>
8600,2,GotoIf($[${CALLERID(num)} = 2002]?50:3)
exten =>
8600,3,GotoIf($[${CALLERID(num)} = 2003]?50:4)
exten =>
8600,4,GotoIf($[${CALLERID(num)} = 2004]?50:5)
exten => 8600,5,GotoIf($[${CALLERID(num)}
= 2005]?50:6)
exten =>
8600,6,GotoIf($[${CALLERID(num)} = 2006]?50:7)
exten =>
8600,7,GotoIf($[${CALLERID(num)} = 2007]?50:8)
exten =>
8600,8,GotoIf($[${CALLERID(num)} = 2008]?50:9)
exten =>
8600,9,GotoIf($[${CALLERID(num)} = 2009]?50:10)
exten =>
8600,10,GotoIf($[${CALLERID(num)} = 2010]?50:11)
exten =>
8600,11,GotoIf($[${CALLERID(num)} = 2011]?50:12)
exten =>
8600,12,GotoIf($[${CALLERID(num)} = 2012]?50:13)
exten =>
8600,13,GotoIf($[${CALLERID(num)} = 2013]?50:51)
exten => 8600,50,Set(CALLERID(num)=2000)
exten =>
8600,51,VoicemailMain(${CALLERID(num)}|s)
exten => 8600,52,Hangup
From: Keith O'Brien
Sent: Sunday, September 24, 2006
12:11 PM
To:
'asterisk-users@lists.digium.com'
Subject: RE: 1.4 Beta 2 Config
Problem
Importance: High
Still no good. Here is what I have now. It looks
like the problem is in my “set” and VoicemailMain statements.
exten => 8600,1,GotoIf($[${CALLERID(num)} = 2001]?50:2)
exten => 8600,2,GotoIf($[${CALLERID(num)} = 2002]?50:3)
exten => 8600,3,GotoIf($[${CALLERID(num)} = 2003]?50:4)
exten => 8600,4,GotoIf($[${CALLERID(num)} = 2004]?50:5)
exten => 8600,5,GotoIf($[${CALLERID(num)} = 2005]?50:6)
exten => 8600,6,GotoIf($[${CALLERID(num)} = 2006]?50:7)
exten => 8600,7,GotoIf($[${CALLERID(num)} = 2007]?50:8)
exten => 8600,8,GotoIf($[${CALLERID(num)} = 2008]?50:9)
exten => 8600,9,GotoIf($[${CALLERID(num)} = 2009]?50:10)
exten => 8600,10,GotoIf($[${CALLERID(num)} = 2010]?50:11)
exten => 8600,11,GotoIf($[${CALLERID(num)} = 2011]?50:12)
exten => 8600,12,GotoIf($[${CALLERID(num)} = 2012]?50:13)
exten => 8600,13,GotoIf($[${CALLERID(num)} = 2013]?50:51)
exten => 8600,50,Set(CALLERID(num)=2000)
exten => 8600,51,VoicemailMain(${CALLERIDNUM}|s)
exten => 8600,52,Hangup
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/2003-b7702ef0",
"0?50:2") in new stack
-- Goto (in-out,8600,2)
-- Executing [EMAIL PROTECTED]:2]
GotoIf("SIP/2003-b7702ef0", "0?50:3") in new stack
-- Goto (in-out,8600,3)
-- Executing [EMAIL PROTECTED]:3]
GotoIf("SIP/2003-b7702ef0", "1?50:4") in new stack
-- Goto (in-out,8600,50)
-- Executing [EMAIL PROTECTED]:50]
Set("SIP/2003-b7702ef0", "CALLERID(num)=2000") in new stack
-- Executing [EMAIL PROTECTED]:51]
VoiceMailMain("SIP/2003-b7702ef0", "|s") in new stack
-- Playing 'vm-login' (language 'en')
[Sep 23 17:56:32] WARNING[20496]: app_voicemail.c:5875
vm_authenticate: Couldn't read username
CALLERID(number) is invalid use CALLERID(num)
[Description]
Gets or sets Caller*ID data on the channel. The allowable
datatypes
are "all", "name", "num",
"ANI", "DNID", "RDNIS".
To:
'asterisk-users@lists.digium.com'
Subject: 1.4 Beta 2 Config Problem
I just upgraded from 1.2.12.1 to 1.4 beta 2 and am having a
problem resolving an issue with the following configuration.
The logic below worked fine in 1.2 but seems to be broken in
1.4 beta 2. The statements 50 and 51 don’t seem to
properly reassign the caller id to “2000” or some other 4 digit
extension. Before I was able to reassign the extension to
say, 2000, and the VoiceMailMain app would drop the user in the correct
mailbox.
Can anyone see what is wrong with the following relative to
1.4 beta 2??
Thank in advance.
exten => 8600,1,GotoIf($[${CALLERID(number)} = 2001]?50:2)
exten => 8600,2,GotoIf($[${CALLERID(number)} = 2002]?50:3)
exten => 8600,3,GotoIf($[${CALLERID(number)} = 2003]?50:4)
exten => 8600,4,GotoIf($[${CALLERID(number)} = 2004]?50:5)
exten => 8600,5,GotoIf($[${CALLERID(number)} = 2005]?50:6)
exten => 8600,6,GotoIf($[${CALLERID(number)} = 2006]?50:7)
exten => 8600,7,GotoIf($[${CALLERID(number)} = 2007]?50:8)
exten => 8600,8,GotoIf($[${CALLERID(number)} = 2008]?50:9)
exten => 8600,9,GotoIf($[${CALLERID(number)} = 2009]?50:10)
exten => 8600,10,GotoIf($[${CALLERID(number)} = 2010]?50:11)
exten => 8600,11,GotoIf($[${CALLERID(number)} = 2011]?50:12)
exten => 8600,12,GotoIf($[${CALLERID(number)} = 2012]?50:13)
exten => 8600,13,GotoIf($[${CALLERID(number)} = 2013]?50:51)
exten => 8600,50,Set(CALLERID(number)=2000)
exten => 8600,51,VoicemailMain(${CALLERIDNUM}|s)
exten => 8600,52,Hangup
-- Executing [EMAIL PROTECTED]:1]
GotoIf("SIP/2002-086a4220", "0?50:2") in new stack
-- Goto (in-out,8600,2)
-- Executing [EMAIL PROTECTED]:2]
GotoIf("SIP/2002-086a4220", "1?50:3") in new stack
-- Goto (in-out,8600,50)
-- Executing [EMAIL PROTECTED]:50]
Set("SIP/2002-086a4220", "CALLERID(number)=2000") in new
stack
-- Executing [EMAIL PROTECTED]:51]
VoiceMailMain("SIP/2002-086
Re: [asterisk-users] RE: 1.4 Beta 2 Config Problem
Unless there's a problem with your "cut & paste", you did not make the
change I proposed. Verified and working here:
exten => 8600,50,Set(CALLERID(num),2000)
exten => 8600,51,VoicemailMain(${CALLERID(num)}|s)
Notice how VoiceMailMain also uses ${CALLERID(num}, not ${CALLERIDNUM}
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[asterisk-users] RE: 1.4 Beta 2 Config Problem
Still no good. Here is what I have now. It looks
like the problem is in my “set” and VoicemailMain statements.
exten => 8600,1,GotoIf($[${CALLERID(num)} = 2001]?50:2)
exten => 8600,2,GotoIf($[${CALLERID(num)} = 2002]?50:3)
exten => 8600,3,GotoIf($[${CALLERID(num)} = 2003]?50:4)
exten => 8600,4,GotoIf($[${CALLERID(num)} = 2004]?50:5)
exten => 8600,5,GotoIf($[${CALLERID(num)} = 2005]?50:6)
exten => 8600,6,GotoIf($[${CALLERID(num)} = 2006]?50:7)
exten => 8600,7,GotoIf($[${CALLERID(num)} = 2007]?50:8)
exten => 8600,8,GotoIf($[${CALLERID(num)} = 2008]?50:9)
exten => 8600,9,GotoIf($[${CALLERID(num)} = 2009]?50:10)
exten => 8600,10,GotoIf($[${CALLERID(num)} = 2010]?50:11)
exten => 8600,11,GotoIf($[${CALLERID(num)} = 2011]?50:12)
exten => 8600,12,GotoIf($[${CALLERID(num)} = 2012]?50:13)
exten => 8600,13,GotoIf($[${CALLERID(num)} = 2013]?50:51)
exten => 8600,50,Set(CALLERID(num)=2000)
exten => 8600,51,VoicemailMain(${CALLERIDNUM}|s)
exten => 8600,52,Hangup
-- Executing [EMAIL PROTECTED]:1]
GotoIf("SIP/2003-b7702ef0", "0?50:2") in new stack
-- Goto (in-out,8600,2)
-- Executing [EMAIL PROTECTED]:2]
GotoIf("SIP/2003-b7702ef0", "0?50:3") in new stack
-- Goto (in-out,8600,3)
-- Executing [EMAIL PROTECTED]:3]
GotoIf("SIP/2003-b7702ef0", "1?50:4") in new stack
-- Goto (in-out,8600,50)
-- Executing [EMAIL PROTECTED]:50]
Set("SIP/2003-b7702ef0", "CALLERID(num)=2000") in new stack
-- Executing [EMAIL PROTECTED]:51]
VoiceMailMain("SIP/2003-b7702ef0", "|s") in new stack
-- Playing 'vm-login' (language 'en')
[Sep 23 17:56:32] WARNING[20496]: app_voicemail.c:5875
vm_authenticate: Couldn't read username
CALLERID(number) is invalid use CALLERID(num)
[Description]
Gets or sets Caller*ID data on the channel. The allowable datatypes
are "all", "name", "num",
"ANI", "DNID", "RDNIS".
To:
'asterisk-users@lists.digium.com'
Subject: 1.4 Beta 2 Config Problem
I just upgraded from 1.2.12.1 to 1.4 beta 2 and am having a problem
resolving an issue with the following configuration. The
logic below worked fine in 1.2 but seems to be broken in 1.4 beta 2.
The statements 50 and 51 don’t seem to properly
reassign the caller id to “2000” or some other 4 digit extension.
Before I was able to reassign the extension to say, 2000, and
the VoiceMailMain app would drop the user in the correct mailbox.
Can anyone see what is wrong with the following relative to
1.4 beta 2??
Thank in advance.
exten => 8600,1,GotoIf($[${CALLERID(number)} = 2001]?50:2)
exten => 8600,2,GotoIf($[${CALLERID(number)} = 2002]?50:3)
exten => 8600,3,GotoIf($[${CALLERID(number)} = 2003]?50:4)
exten => 8600,4,GotoIf($[${CALLERID(number)} = 2004]?50:5)
exten => 8600,5,GotoIf($[${CALLERID(number)} = 2005]?50:6)
exten => 8600,6,GotoIf($[${CALLERID(number)} = 2006]?50:7)
exten => 8600,7,GotoIf($[${CALLERID(number)} = 2007]?50:8)
exten => 8600,8,GotoIf($[${CALLERID(number)} = 2008]?50:9)
exten => 8600,9,GotoIf($[${CALLERID(number)} = 2009]?50:10)
exten => 8600,10,GotoIf($[${CALLERID(number)} = 2010]?50:11)
exten => 8600,11,GotoIf($[${CALLERID(number)} = 2011]?50:12)
exten => 8600,12,GotoIf($[${CALLERID(number)} = 2012]?50:13)
exten => 8600,13,GotoIf($[${CALLERID(number)} = 2013]?50:51)
exten => 8600,50,Set(CALLERID(number)=2000)
exten => 8600,51,VoicemailMain(${CALLERIDNUM}|s)
exten => 8600,52,Hangup
-- Executing [EMAIL PROTECTED]:1]
GotoIf("SIP/2002-086a4220", "0?50:2") in new stack
-- Goto (in-out,8600,2)
-- Executing [EMAIL PROTECTED]:2] GotoIf("SIP/2002-086a4220",
"1?50:3") in new stack
-- Goto (in-out,8600,50)
-- Executing [EMAIL PROTECTED]:50]
Set("SIP/2002-086a4220", "CALLERID(number)=2000") in new
stack
-- Executing [EMAIL PROTECTED]:51]
VoiceMailMain("SIP/2002-086a4220", "|s") in new stack
-- Playing 'vm-login' (language 'en')
[Sep 23 11:42:25] WARNING[14722]: app_voicemail.c:5875 vm_authenticate:
Couldn't read username
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Re: [asterisk-users] PRI Backup
I don't know if this may at sometime help mr Wood, but BT, with their ISDN30* actually offer something called Site Assurance - the problem is that it does not automatically fail over, and according to the last memo I read - failover takes about 1 hr. A problem is that, due to outsourcing, product ranges, size issues, etc, a lot of people on BT's frontline are not really keyed up to their product offerings. Who knowns, maybe the failover process has been automated at this point in time. Conrad Wood <[EMAIL PROTECTED]> wrote: >> making the call. I guess I could just add the call route to the other> campus just below the my default call route. So if the primary call> route fails, it will just go to the next line being the other campus.>That's precisely what I do with the main route out on ISDN, if that fails, it switches over to various voip providers and even down to a bluetooth enabled mobile ;).it works quite allright for outgoing calls.I believe for incoming calls you need to persuade your isdn supplier do forward the call to ISDN-B if ISDN-A is hosed.Here in UK I couldn't persuade BT to do so yet ;(Conrad___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All New Yahoo! Mail Tired of [EMAIL PROTECTED]@! come-ons? Let our SpamGuard protect you.___ --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] Asterisk+Astbill
how do I integrate asterisk with asbill ? ___ --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] DSL router with integrated SIP proxy?
You could take a WRTSL54gs, install openwrt then openser David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Kennedy Envoyé : 24 septembre 2006 08:47 À : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] DSL router with integrated SIP proxy? On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: > Does anyone here know of an ADSL router with integrated SIP proxy? Netscreen 5GT ADSL, it has what's called an ALG (application layer gateway) and it does indeed support SIP. Full featured firewall etc too. Steve p.s Hi Brian :) -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
2 cents I would not mind paying a reasonable price for a single port BRI but buying a Quad-BRI to get a stable installation is a bit too much for most home installations. Then I will probably start using the old Digital->Analog adapter and use a TDM card. But I don't understand why it shouldn't work with a HFC-S bri card. The way I understand it is that HFC-S cards are quite dumb and a lot of work needs to be done with the main cpu. The ISDN signalling is quite sensitive to timing and if the main cpu is busy it's not going to be happy. It also appears that capi/i4l/misdn have a latency far higher than what is useful for asterisk. It is aimed at general-purpose use, including data transfer. At least, that was my experience. I tried visdn and the results where very promising but it doesn't seem to be quite ready yet. I suspect it's architecture will be much saner than bristuff patches. I also think you get what you pay for and I don't use hfc based isdn cards in production any more. Having said that, a small home installation isn't quite the same as a 30 user office environment. My home-pbx for example is quite happy reloading asterisk+zaphfc every night. Of course not something I'd accept in a production environment, but that's probably not what HFC-s cards are aimed at either, right? Conrad ___ --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] PRI Backup
making the call. I guess I could just add the call route to the other campus just below the my default call route. So if the primary call route fails, it will just go to the next line being the other campus. That's precisely what I do with the main route out on ISDN, if that fails, it switches over to various voip providers and even down to a bluetooth enabled mobile ;). it works quite allright for outgoing calls. I believe for incoming calls you need to persuade your isdn supplier do forward the call to ISDN-B if ISDN-A is hosed. Here in UK I couldn't persuade BT to do so yet ;( Conrad ___ --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] DSL router with integrated SIP proxy?
On 24 Sep 2006, at 13:47, Steve Kennedy wrote: On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: Does anyone here know of an ADSL router with integrated SIP proxy? I use soekris boxes with openbsd on a flash card and a lot of scripting to gather statistics on all sorts of stuff. works very well too and gives all the stats one can wish for ;) conrad ___ --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] PRI Backup
Ps. Mr Beck, if you do decide to, at sometime, try to take the TDMoE route - these pages give good pointers: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1320.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TDMoE Yet, IMHO, TDMoE is used to encapsulate raw TDM (say ISDN) traffic in Ethernet for transportation, and is best suited for bridging/backhauling. An example would be, PSTN->PRI->(TDMoE)->IPNetwork->(TDMoE)->PRI->PSTN flows as is currently used by many telcommunication companies. I am, however, very greatful to Massimiliano for bringing this up, as I did not know that Asterisk could take care of the mux/demux + stringent timing requirements related to TDM transfer in this manner. I would however like to know how Asterisk "generally" performs when compared to other TDMoE provisions such as those sold by RAD or indeed CESoP - if anyone has any answers, your input would be more than welcome. Regards Bayo adebayo omo-dare <[EMAIL PROTECTED]> wrote:There is the very great possibility that employing TDMoE in this environment will introduce new levels of complexity in to the network. And though TDMoE, in itself, is fantastic, IMHO - long haul technology, it may also be considered to be out of scope and limiting/expensive, most especially considering he already employs the type of network many dream of. In terms of distributing inward bound PRI calls across the two sites, your telecom company takes care of those details and forwards calls to the other when the former is seen to be busy. [They possibly call this something like Diversion/Forwarding on Busy] - speak to your provider. In terms of distributing outgoing VoIP->PRI->PSTN calls - Off the top of my head, and there may/should be much better ways, you may/should be able to introduce a global count variable and a GotoIf(...) in the dial plan. Hope this, in some manner, helps Bayo Massimiliano Stucchi <[EMAIL PROTECTED]> wrote:On 200906, 16:00, Forrest Beck wrote:> > I am looking to see if anyone has a dial plan setup to use a secondary> PRI. We have two campuses, each with it's own PRI (for telco going to> a single span digium card) and a 10MB fiber link between the two for> data. All calls are transfered between the two campuses via the 10MB> data line and outgoing calls are made on the campus'es PRI. I am> loking to see if there is a way to tell the server if one PRI is full> (all 23 channels are in use) or not available to try routing the call> through the other campus and it's PRI. Anyone doing this already? I> am not sure how to have asterisk check to see if a PRI is down for> making the call. I guess I could just add the call route to the other> campus just below the my default call route. So if the primary call> route fails, it will just go to the next line being the other campus.I would try using TDMoE, and duplicating the PRIs over the two machines,so that:Machine 1 handles PRI A and has B as, say group2Machine 2 handles PRI B and has A as, say group2I don't know if you can run TDMoE over your fiber connection, but I wasjust here for a suggestion.Ciao-- Massimiliano Stucchi, CTO & Director of OperationsWillyStudios.com - IT Consulting, Web and VoIP Services[EMAIL PROTECTED] | Tel (+39) 0244417203 | Fax (+39) 0244417204IT-20040, Carnate (Milano), via Carducci 9___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Try the all-new Yahoo! Mail . "The New Version is radically easier to use" The Wall Street Journal___ --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] dialplan for confrencing
Can anyone give me dial plan for thirdparty confrencing without channel redirect. I think channel redirct command is not supported in asterisk now. Thanks Imthiyaz mail2web - Check your email from the web at http://mail2web.com/ . ___ --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: [hylafax-users] [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Hello, # cat /proc/interrupts CPU0 0: 2087872259IO-APIC-edge timer 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 14: 18440124IO-APIC-edge ide0 15:4456445IO-APIC-edge libata 169:4878102 IO-APIC-level eth0 177: 2086847525 IO-APIC-level wctdm24xxp 185: 2086810653 IO-APIC-level wct4xxp NMI: 0 LOC: 2087921792 ERR: 0 MIS: 0 zttest is often on 99.975586% with final result: --- Results after 67 passes --- Best: 99.987793 -- Worst: 99.951172 -- Average: 99.973764 Where should I find good web pages for tuning? I had found this and using now: setpci -v -s 04:06.0 latency_timer=ff setpci -v -s 04:08.0 latency_timer=ff As I remember it's give higher priority for Digium cards. The machine is an IBM eServer x206 with P4 2.66 GHz processor. bye, Zsolt On 9/23/06, Ma Zhiyong <[EMAIL PROTECTED]> wrote: zttest don't disturb your other active calls. when you have 10-20 calls maybe got a better score. also cat /proc/interrupt and 'lspci -vb' to find any IRQ interrupt on your system. 2006/9/22, Artifex Maximus <[EMAIL PROTECTED]>: > Hello, > > On 9/21/06, Lee Howard <[EMAIL PROTECTED]> wrote: > > Artifex Maximus wrote: > > > > > Everything is fine when caller use ECM but when ECM isn't in use I > > > often got unusable incoming faxes (much often that it should be). But > > > when I switch back to fax machine that receive faxes perfectly (at > > > least no visible error). > > The fax machine itself uses ECM, undoubtedly. > That's unfortunately not the case. The remote doesn't asks for ECM so > that's disabled or missing on that machine. In this situation fax > machine is produce better output and I don't know why. Might a better > DSP algo? > > > If callers that have > > quality problems with IAXmodem+HylaFAX don't have problems with the fax > > machine, then that strongly indicates that something is wrong with your > > Asterisk setup... corrupting the audio. Usually this is due to resource > > constriction of the Zap device, zttest scores less than 99.98% is > > indicative of that situation. > I don't find any info that zttest is destructive or not on an active > system. I mean that currently active calls are disturbed or not while > zttest running. I can't stop system now. I look into zttest source and > find that zttest is using /dev/zap/pseudo but I don't know this > 'pseudo' channel is related to any 'real' channel or not. > > > > Where should be the problem? Is there any solution for improving > > > quality? Any tuning in Asterisk or Hylafax? As I see some people use > > > slinear for iaxmodem and some user use alaw. Which is better? > > There is no functional difference between using uLaw, alaw, or > > slinear... except that using slinear reduces the need for conversion... > > and thus possibly lessens CPU usage very slightly. > I see. I leave it on slinear. > > bye, > Zsolt ___ --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] UK Male English Voices
On Fri, Sep 22, 2006 at 02:56:39PM +0100, Will Tatam wrote: > Steve Kennedy wrote: > >I'd like to announce that the UK Male English Voices are now up on > >http://www.tel.net/ [snip] > The website appears to be down Yup, did an upgrade on Fri and something went wrong - will be fixed tomorrow. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] DSL router with integrated SIP proxy?
On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: > Does anyone here know of an ADSL router with integrated SIP proxy? Netscreen 5GT ADSL, it has what's called an ALG (application layer gateway) and it does indeed support SIP. Full featured firewall etc too. Steve p.s Hi Brian :) -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] running ooh323 on asterisk-1.14beta2
Dear Sirs, I tried to run asterisk-1.14beta2 + ooh323 on connection to AddPac, which is the following (only h323 important stuff): AddPac 1100C version 8.234 voice service voip fax protocol t38 redundancy 0 fax rate 2400 h323 call start fast h323 call channel early h323 call tunnel enable inband-ringback-tone announcement language english however, asterisk complains on unknown RTCP packets (code 207) and sound is not transmitted. I also tried similar connection to Cisco 3530 using faststart, it performs much better than earlier version of ooh323, which is supplied with asterisk-1.12. I can provide any debug/log/config related to that setup. Can anybody help me with AddPac ? С уважением, Илья Шипицин технический директор рекламная группа PARAMON 454080,Россия, Челябинск,пр.Ленина,78-Б телефон: 8 912 793-96-21 mailto: [EMAIL PROTECTED], icq: 177725537 www.paramon.ru ___ --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] PRI Backup
There is the very great possibility that employing TDMoE in this environment will introduce new levels of complexity in to the network. And though TDMoE, in itself, is fantastic, IMHO - long haul technology, it may also be considered to be out of scope and limiting/expensive, most especially considering he already employs the type of network many dream of. In terms of distributing inward bound PRI calls across the two sites, your telecom company takes care of those details and forwards calls to the other when the former is seen to be busy. [They possibly call this something like Diversion/Forwarding on Busy] - speak to your provider. In terms of distributing outgoing VoIP->PRI->PSTN calls - Off the top of my head, and there may/should be much better ways, you may/should be able to introduce a global count variable and a GotoIf(...) in the dial plan. Hope this, in some manner, helps Bayo Massimiliano Stucchi <[EMAIL PROTECTED]> wrote:On 200906, 16:00, Forrest Beck wrote:> > I am looking to see if anyone has a dial plan setup to use a secondary> PRI. We have two campuses, each with it's own PRI (for telco going to> a single span digium card) and a 10MB fiber link between the two for> data. All calls are transfered between the two campuses via the 10MB> data line and outgoing calls are made on the campus'es PRI. I am> loking to see if there is a way to tell the server if one PRI is full> (all 23 channels are in use) or not available to try routing the call> through the other campus and it's PRI. Anyone doing this already? I> am not sure how to have asterisk check to see if a PRI is down for> making the call. I guess I could just add the call route to the other> campus just below the my default call route. So if the primary call> route fails, it will just go to the next line being the other campus.I would try using TDMoE, and duplicating the PRIs over the two machines,so that:Machine 1 handles PRI A and has B as, say group2Machine 2 handles PRI B and has A as, say group2I don't know if you can run TDMoE over your fiber connection, but I wasjust here for a suggestion.Ciao-- Massimiliano Stucchi, CTO & Director of OperationsWillyStudios.com - IT Consulting, Web and VoIP Services[EMAIL PROTECTED] | Tel (+39) 0244417203 | Fax (+39) 0244417204IT-20040, Carnate (Milano), via Carducci 9___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine___ --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] Segmentation fault on Asterisk startup: res_config_mysql.so problem?
On 16:04, Sun 24 Sep 06, kjcsb wrote:
> When Asterisk starts I get a Segmentation fault
> /usr/sbin/safe_asterisk: line 40: 30548 Segmentation fault (core
> dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} Asterisk ended with exit status 139
> Asterisk exited on signal 11.
>
> If I remove /usr/lib/asterisk/modules/res_config_mysql.so Asterisk starts
> normally.
>
> tail /var/log/asterisk/full.log
> Sep 24 15:46:05 VERBOSE[30608] logger.c: == Parsing
> '/etc/asterisk/res_mysql.conf': Sep 24 15:46:05 VERBOSE[30608] logger.c:
> == Parsing '/etc/asterisk/res_mysql.conf': Found
> Sep 24 15:46:05 WARNING[30608] res_config_mysql.c: MySQL RealTime: No
> database socket found, using '/tmp/mysql.sock' as default.
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Host:
> 127.0.0.1
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Port: 3306
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime User: root
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Password:
> password
>
> vi /etc/asterisk/res_mysql.conf
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = root
> dbpass = password
> dbport = 3306
> ;dbsock = /var/lib/mysql/mysql.sock
>
> If I uncomment the dbsock line I get the same result (although the database
> socket warning is not displayed in the log file).
>
> I am using MySQL for CDR logging so I don't think it's a MySQL problem.
>
> Asterisk 1.2.12.1
> Asterisk addon 1.2.4
>
> When I install Asterisk I receive a warning:
> Your Asterisk modules directory, located at /usr/lib/asterisk/modules
> contains modules that were not installed by this version of Asterisk.
>
> However I cleared out the /usr/lib/asterisk/modules directory before make
> clean && make && make install for both add-ons and asterisk so I'm a bit
> mystified by that.
>
Did you do a make && make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
"Why is it drug addicts and computer afficionados are both called users?"
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