Re: [Asterisk-Users] New voiceovers for Allison Smith: submit today
I'm a native Portuguese (European) speaker but I really haven't been following this thread. Is there anything I can help out with? (Reply directly to my email for a quicker response). Regards, Leandro Paul Davidson wrote: Unfortunately, I do not have the correct pronounciations- but there are some sounds missing in say.c, for at least Portuguese: pt-ah.gsm pt-ao.gsm pt-de.gsm pt-e.gsm pt-ora.gsm pt-meianoite.gsm pt-meiodia.gsm pt-sss.gsm From what I can tell, they've been missing from the main repository for a few years, yet have been referenced in say.c for quite a while. Since I'm not a native portuguese-speaker, I'm entirely the wrong person to give a pronounciation gude here- but perhaps one of the Brazillian subscribers can? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: 20 July 2005 23:55 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] New voiceovers for Allison Smith: submit today I'm sending in a set of voiceover requests to Allison Smith this afternoon. I haven't kept up with the -users list to know if there is someone keeping track of this stuff any more... We only have a few phrases for her to record, and if anyone has applications which require Allison's voice for the "asterisk-sounds" repository, let me know. I'll be sending this in around 22:00 PDT today, so act fast. Please format the requests in the style: %filename%text-to-speak example: %auth-incorrect.gsm%Login incorrect. Please enter your password followed by the pound key. Any pronunciation keys should be in-line, inside of [brackets]. Please email directly to me. JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Interface with mobile phone
Hey guys, We have a couple of Nokia 32 GSM units left over from a client's * installation. The units than can be hooked up to a FXO or FXS (it's got 2 ports) and work pretty well in production connected to a FXO port. Don't try to bridge calls using 2 of these though. If anyone is interested in adquiring one/both of these drop me a line. More info: http://www.nokia.com/nokia/0,8764,38621,00.html Leandro Juraj Bednar wrote: >Hello, > > > >>There's this device called VoiceBlue GSM gateway. >>It talks gsm on one side and SIP on the other side. >>Have a look at: >>http://www.voip-info.org/tiki-print.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX >> >> > >yep, but it is very expensive, I found. Even cellphone + cellsocket + >FXO card would be cheaper than this. > >I want to do the same thing, I will try to use chan_bluetooth, but >its' svn repository is unaccessible right now :(. > > Juraj. >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe wcfxo fails.
Hey Tim, I'll be glad to help you out if I am able to.. but I honestly don't recall which thread you are talking about. Maybe if you refresh my mind and/or explain your problem? Leandro Tim King wrote: I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer & Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?
Steve Underwood wrote: > Robert Rozman wrote: > >> Hi, >> >> I'm getting unreliable dtmf recognition (it works fine for 4-5 >> digits, errors (duplicates) on more), when transferred inband from >> gsm gateway to NT port of quadbri under bristuffed Asterisk. >> >> Since Asterisk is claimed to have good dtmf recognizer, I suspect >> there are some settings to workarouned... I've tried dtmf relax, but >> didn't help, so I suspect gain settings >> >> Is there any other possible cause of unreliable dtmf inband >> recognition ? Where can I set gain on voice channel (I guess majority >> of settings under bristuff in zaptel.conf are dummy) ? >> >> Any other advice on this problem or similar experience ? >> >> Thanks in advance, > > > I kind of amazed if works at all when getting DTMF out of a GSM phone. > You really shouldn't expect it to. We have sucessfully "read" incoming DTMF from: a) Nokia32 Analog GSM connected to TDM400 (had to use relaxdtmf with chan_zap) b) Ateus BRI ISDN GSM connected to AVM Fritz (had to patch chan_capi 0.3.5 to support relaxdtmf) Question (I'm from a software eng. background, not telco): What would be the reason for not receiving DTMF from a GSM phone/gateway? Do you have the time to explain why? (I'm really interested in learning :) Thanks, Leandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LookupCIDName on outgoing calls
I'm trying to use LookupCIDName to tag outgoing calls on my CDRs but it seems that application only tags incoming calls? Any sugestions? Leandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ISDN 4 BRI card for UK
[snip] Sorry Tony, but he was talking about the UK :) 4 Fritz! or HFC-S card do the job. A 4 port HFC card would still be cheaper than the Eicon. /Martin Hi! How did you manage to get 4 Fritz cards in the same box? Could you give details on what kernel and kmodules you used? Which asterisk channel are you using? chan_capi? chan_misdn? Thanks, Leandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Measure the Signal of Zap
Oh.. the voltage values are dumped in syslog!
Leandro Morgado wrote:
Damian Minkov wrote:
Is there a way to measure the signal of the connected line on the FXO
port ( without the help of digital oscilloscope )
Yes there is. But you need to edit the source code of wcfxs (for the
TDM400 card). There is a bit of code similar to:
#if 0
-->some debug messages here regarding voltage values
#endif
You can simply replace it for
a) if 1
This will always print out voltage values... LOTS of them!
or
b) if (debug>1) {}
This will print out voltage values when you "modprobe wcfxs.c debug=2"
There should be similar bits of code in the X100P driver.
Leandro
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Re: [Asterisk-Users] Measure the Signal of Zap
Damian Minkov wrote:
Is there a way to measure the signal of the connected line on the FXO
port ( without the help of digital oscilloscope )
Yes there is. But you need to edit the source code of wcfxs (for the
TDM400 card). There is a bit of code similar to:
#if 0
-->some debug messages here regarding voltage values
#endif
You can simply replace it for
a) if 1
This will always print out voltage values... LOTS of them!
or
b) if (debug>1) {}
This will print out voltage values when you "modprobe wcfxs.c debug=2"
There should be similar bits of code in the X100P driver.
Leandro
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Re: [Asterisk-Users] asterisk sounds
Josiah Bryan wrote: On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote: Hello all, I am looking for a list of all available sound files for asterisk and a transcription of their content, so that I can have someone translate them into portuguese. I vaguely remeber reading some file in my server that had a list of all the sound files and their transcripts...i just spent about 20 minutes looking for it in the /usr/src/asterisk CVS tree that I checked out - cant seem to find it off hand. Any body have any idea what that file is? sounds.txt :) -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing CAPI
Hi, I've used the Fritz AVM PCI card with Junghanns.net chan_capi and it's working great. It's never crashed or given problems, although I have a low call volume (at most 50 calls a day). The setup was not straight forward (fritz drivers compilation, version matching, etc) but it wasn't very dificult with help from the very same links you gave (wiki and junghanns docs). Maybe it's a problem with your ISDN card? I've tried 2 other cards and just couldn't get them to work. The Fritz works great though! Leandro Damian Funnell wrote: Hi there, We recently did our first * install with CAPI and we found the levels of support (and general knowledge) within the community seriously wanting. In fact, we found things so bad that I would caution against using CAPI unless you are feeling particularly game and confident in your abilities to fix problems, as you are likely to find it very difficult to get help if you need it. Out of the half dozen or so help requests that I or my colleagues posted to this forum or to the #asterisk IRC channel, for example, we didn't receive a single helpful response. Not one. Not that there wasn't anyone who was willing to help, but there just didn't seem to be anyone around who was using CAPI in anger. We originally chose CAPI over ISDN4Linux because of the commercial support that was supposedly available through junghanns.net (CAPI also provides a better feature set than ISDN4Linux, but we don't use any of the additional features, so this wasn't a consideration for us), but when we called upon junghanns.net for support it took them so long to respond that we needn't have bothered (we had stumbled across a fix ourselves by the time we got a response from them). If this hasn't scared you off then check out the documentation at http://www.junghanns.net/asterisk/ and the sample files/readme that come with the CAPI source. There is also a fairly good configuration guide at http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI and the CAPI readme is reproduced at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme. Drop me a mail at damian dot funnell at fff dot co dot nz if you would like me to send you a copy of our conf files so you can see how we're using it. Right now we are trying to diagnose a problem where the voice channels over CAPI fall apart a few times per day, resulting in all external calls having to be terminated. We don't know if this problem is CAPI related, but predictably we haven't been able to find anyone in the community who can help us figure it out. Best regards, Damian. [EMAIL PROTECTED] wrote: Hi! I can't find any instructions of installing capi and chan_capi. Do you know any site with instructions or can you give me step by step help with this. Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phonecell + wildcard FXO (DTMF problems)
jafar mohammed wrote: Hi, I purchashed a Telular Phonecell Fixed Cellular Terminal. I hooked it up to my wildcard fxo card. I can receive calls and these calls are passed on to the Asterisk Calling Card application. My problem is that i can't get DTMF to work properly. If a pin number is 484443543639 i get 484333544336639. how can i sort out this problem. Please would like ur urgent assistance. I had the same problem. Try using relaxdtmf=yes in your zapata.conf . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisk to GSM
Jean-Michel Hiver wrote: Hi List, I was wondering if there was any device I could use to connect * to GSM networks. I don't need much capacity, maybe 2-4 GSM channels. As usual, cheap is better :-) I've used Nokia 32 with a TDM400 FXO. It works reasonably well but has some anoying "features", like taking 5-10 seconds to establish a call, hangup detection problems, cdr records always being answered, etc, etc. It's doable but not perfect. Any tips on this? I haven't tried ISDN GSM Terminals but do have fixed-land ISDN. ISDN is so much more reliable (faster setup, proper cdr and billing seconds, better voice quality) that I would consider a ISDN GSM terminal. Try searching google for quasar smartcell. A Voip-GSM terminal sounds even better. No need for FXO, ISDN, etc. And you could manage it remotly. The Ateus Voice Blue looks good but I haven't had a change to try it out. Google around.. you find nice stuff like: http://www.thehightechstore.com/cellinterface.html Let me know about your experiences. Regards, Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
On Thu, 2004-11-25 at 00:10, Jose Hernandez wrote: > I added the rules to udev permissions file and changed zaptel.conf, > restarted. > I still get the same error; > > [EMAIL PROTECTED] frank]# ztcfg -vv > Zaptel Configuration > == > Channel map: > > Channel 01: FXS Kewlstart (Default) (Slaves: 01) > Channel 02: FXO Kewlstart (Default) (Slaves: 02) > > 2 channels configured. > > Notice: Configuration file is /etc/zaptel.conf > line 5: Unable to open master device '/dev/zap/ctl' > > zaptel.conf > fxsks=1 ;X100P > fxoks=2 ;TDM400, TDM10B (1 FXS port) > loadzone=us > defaultzone=us > I don't know if it makes a difference but try putting loadzone and defaultzone before the fxXks declarations. Also be careful with spacing. I remember once I had problems because a config file (don't remember if it was zapata.conf or zaptel.conf) had (or had no) spaces when I "declared" the channels. Little parsing glitches! :) > This is the output from "ls -al za*" in /dev > crw--- 1 root root 196, 1 Nov 23 15:33 /dev/zap1 > crw--- 1 root root 196, 254 Nov 23 15:33 /dev/zapchannel > crw--- 1 root root 196, 0 Nov 23 15:33 /dev/zapctl > crw--- 1 root root 196, 255 Nov 23 15:33 /dev/zappseudo > crw--- 1 root root 196, 253 Nov 23 15:33 /dev/zaptimer > > Are these names correct? > The error has /dev/zap/ctl but in /dev I have /dev/zapctl As the permissions are now it will only work if * runs as root (which is a bad idea). Do a 'ps aux|grep asterisk' and find out which user asterisk is running as. Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
On Wed, 2004-11-24 at 13:53, Dave Cotton wrote: > On Wed, 2004-11-24 at 13:45 +0000, Leandro Morgado wrote: > > Jose Hernandez wrote: > > > > >I installed TDM400P and X100P pci cards in a system running mandrake 10.1 > > >official, kernel 2.6.8.1-12mdksmp. > > > This is not udev up to its tricks again? You are right! If his distro is using udev it might dynamically create the zap devices with the wrong set of permissions. I am not using udev (debian 2.4) but it makes sense that udev would allow you to specify the permissions to use when creating devices! Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
Jose Hernandez wrote: I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [EMAIL PROTECTED] asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl' [EMAIL PROTECTED] asterisk]# asterisk -vvvcg ... == Parsing '/etc/asterisk/zapata.conf': Found Nov 22 21:16:11 WARNING[14643]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/channel': No such file or directory Maybe the zaptel devices in /dev were not created properly. If they are there, make sure that asterisk is running as a user that has permission to read the /dev/zaptel devices. Hope that helps! Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mISDN & kernel 2.6.9
Thomas Jagoditsch wrote: hi christiaan. Christiaan Brink schrieb: [snip] sorry, but i use chan_capi with the avm B1 card (this is an active one, different then a A1/fritzcard), see my posting. i see no way to use a hfc-based card or an avm A1/fritz with chan_capi. so IMHO you need not chan_capi but chan_misdn (or alternatively zaphfc - which despite trying hard i couldnt compile with the 2.6.9 kernel :-( ) It is possible to use an AVM Fritz (in TE mode) with chan_capi. I've done it myself and it works great. It's documented pretty well in the wiki. You'll need to compile the kernel drivers from AVM and have capi support in the kernel for it to work. I could not get the Fritz working with hisax (in fact I had to remove hisax!) . Again, check the wiki.. It's all there. :) Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Adam Goryachev wrote: On Thu, 2004-11-18 at 05:43, Leandro Morgado wrote: Matt Riddell wrote: Leandro Morgado wrote: Although, I still think that there is some kind of incompatibility or battery drop timing problem between Asterisk and the Nokia 32. I wish I knew more about telecomms and wcfxs.c to fix it! Thanks mate! (I know you're not Aussie, but I am and consider you NZ folk as our friendly neighbours :) Hi there, just wondering where you sourced these GSM terminals from? I've been thinking about this for a while, but always assumed they'd be pretty scarce in Australia... Hey Adam, I am an Australian citizen but as I said before in this thread, I am located in Portugal now. GSM Terminals are widely available here. I've been able to get my hands on Nokias 32 and also Quasar Smartcells (both analog and ISDN-BRI ones). The analogue 1 channel (SIM card) GSM terminals are around 400-500€. At the moment I'm considering using some GSM-VoIP gateways (from these ppl http://www.2n.cz/ ) but I still have to find out about pricing and availability. Leandro Thanks, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Matt Riddell wrote: Leandro Morgado wrote: Correction! I do need busydetect (i had forgotten to comment it out) to detect hangups. I'm not as familiar with this Smartcell GSM terminal but I dont think it drops the battery when call are hung up (in /var/log/messages voltage stayed at 9V during the call. After I hung up asterisk detects the busytone and goes onhook sending it to 50V.) Thought you might be interested in this patch added to CVS head yesterday: http://www.sineapps.com/news.php?rssid=327 Excerpt: This patch adds code the chan_zap.c file, to allow it to pick up the Polarity Reversal events and process them to indicate the remote end has disconnected. The code is controlled by two new options in the zapata.conf file, and by default, it is disabled. Yes, thanks. I do read Asterisk News. It's a good source of info! :) I'll give it a try sometime soon. These Nokias 32 do have the option of reversing polarity for X ms to indicate a call is connected or they can reverse polarity for the duration of the call. It's nice to see development is being done in zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
On Wed, 2004-11-17 at 18:43, Leandro Morgado wrote: [SNIP] > Well.. i'm just leaving this but the mailing list archives. It really > must be some kind of Disconnect Supervision incompatibility between the > Nokia 32 GSM and Asterisk. Maybe asterisk doesnt like the duration that > the Nokia drops the battery for. > > Anyway, I just tried using Kewlstart (and therefore Disconnect > Supervision) with a Quasar Smartcell 111L and it works flawlessly. I > don't need busydetect and voltage values are reported as expected! > Correction! I do need busydetect (i had forgotten to comment it out) to detect hangups. I'm not as familiar with this Smartcell GSM terminal but I dont think it drops the battery when call are hung up (in /var/log/messages voltage stayed at 9V during the call. After I hung up asterisk detects the busytone and goes onhook sending it to 50V.) [SNIP] Leandro Morgado > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Bragaredes - I.R.I. Telecomunicações, Lda. Rua Padre Freitas, Nº 106 Fracção A, 4700 - 283 Braga, PORTUGAL Cell: +351 962415514 Phone: +351 253 625 399 Fax: +351 253 331 349 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Matt Riddell wrote: Leandro Morgado wrote: Although, I still think that there is some kind of incompatibility or battery drop timing problem between Asterisk and the Nokia 32. I wish I knew more about telecomms and wcfxs.c to fix it! :-) You and I both! Well.. i'm just leaving this but the mailing list archives. It really must be some kind of Disconnect Supervision incompatibility between the Nokia 32 GSM and Asterisk. Maybe asterisk doesnt like the duration that the Nokia drops the battery for. Anyway, I just tried using Kewlstart (and therefore Disconnect Supervision) with a Quasar Smartcell 111L and it works flawlessly. I don't need busydetect and voltage values are reported as expected! However, seeing as you say it can make multiple tones, it may be easiest to set the Nokia to create USA tones, and then in zaptel.conf put loadzone=us, defaultzone=us and in zapata.conf busydetect=yes and busycount=10. Hum.. USA is not on the list of supported countries. Most european countries are, so I just randomly choose France (after hearing about nightmares with CLI in UK mode!) seeing that zaptel.conf supports defaultzone=fr . USA is listed as [us]. I meant that the Nokia 32 doesn't list USA in the country setting options. :) Although this is really a work-around to the problem (the real problem is still out there) I thank you very much for being so helpful! No problems, let us know if you have any other issues. Now all the issues are solved or worked around. Matt, I would really like to thank you for the help you provided. You got me out of a tightspot. :) Let me know if I can ever retribute the favour. Thanks mate! (I know you're not Aussie, but I am and consider you NZ folk as our friendly neighbours :) Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Matt Riddell wrote: You can set it to detect hangups via tones in zapata.conf with the busydetect=yes and busycount=10 entries. From what I was able to figure out from my the logs, asterisk will not do busytone hangup detection when using Kewlstart. I guess Kewlstart expects a battery drop to indicate hangup so I tried Groundstart (which failed to load) and Loopstart. Now when I use Loopstart asterisk doesn't seem to detect the hangup using the battery drop but it does eventually detect it. My guess is that it only uses busydetect with Loopstart signalling. With fxs_ls and the busydetect=yes , busycount=10 voltages values after the gsm signals hangup are correctly indicated as being 48V. It's not as clean as a battery drop (Disconnection Supervision) but hey, at least it works! :) Although, I still think that there is some kind of incompatibility or battery drop timing problem between Asterisk and the Nokia 32. I wish I knew more about telecomms and wcfxs.c to fix it! The changes in zaptel.conf will just change what tones (i.e. Hz) and what cadences (i.e. ms) are used to detect hangup via tones. The opermode switches should change the line impedance etc of the module. By changing the opermode to FRANCE I was trying to affect the electrical characteristics of the FXO module, hoping that the correct 48v would show up in the log files when asterisk hangs up. Seeing that it is now in "busytone hangup detection" mode, the values in zaptel.conf should have a influence on how it does the detection. The recordings I am after are of the hangup tones that the Nokia is producing. Hum.. any idea how I can record these busy tones? Or is this just a waste of time, seeing that asterisk correctly hangs up based on busy tones? However, seeing as you say it can make multiple tones, it may be easiest to set the Nokia to create USA tones, and then in zaptel.conf put loadzone=us, defaultzone=us and in zapata.conf busydetect=yes and busycount=10. Hum.. USA is not on the list of supported countries. Most european countries are, so I just randomly choose France (after hearing about nightmares with CLI in UK mode!) seeing that zaptel.conf supports defaultzone=fr . Although this is really a work-around to the problem (the real problem is still out there) I thank you very much for being so helpful! Thanks, Leandro Morgado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Matt Riddell wrote: Leandro Morgado wrote: [EXTENSIVELY SNIPPED] zaptel.conf: loadzone=fr defaultzone=fr zapata.conf: --- busydetect=yes busycount=7 [EXTENSIVELY SNIPPED] 1. Does the Nokia 32 GSM provide you with a hangup tone? (I.E. beep beep beep once it has hung up) Yes it does. If I connect a regular phone to the Nokia, I hear a "beep beep beep" after the caller on the cell phone has hungup. If I don't put the phone back onhook an "alarm" sound will ring, just like on a regular land line, and the line will stay busy (this is exactly what asterisk is not doing properly). Notice that the Nokia GSM Terminal supports "Disconnect Supervision" by dropping the battery for X ms (X is configurable and is currently at 500ms). The FXO module does correctly identify this as indication that the call has hung up. I'm pretty sure the FXO module is detecting hangups through battery drop and not by listening for busy tones, but I could be wrong here. 2. You have specified France as the zone, are these the tones that the Nokia unit produces? I can set up the Nokia to use a range of different European countries. I have tried various combinations in country settings, both on the Nokia, in zaptel.conf and when I do "insmod wcfxs opermode=XXX". These include FCC, TBR21, FRANCE and PORTUGAL (where I am located). Although I tested various country settings it didn't seem to make much different in the voltage values spitted out in /var/log/messages. I have not tested for different "busy" tones according to country settings but I will do so at the next opportunity and tell you how it goes. If you have some audio files from voicemail of the busy signal, you can send them to me and I can create a custom indications.conf entry for you (assuming it is different from all other countries in indications.conf). I'll get some recordings for you tomorrow when I'm at the office. I'm located in Portugal but that should not have an influence in the way the Nokia 32 works. Let me know. Thanks for all your help! Leandro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TMD400 FXO <-> Nokia 32 GSM (Hangup Problems)
Hi, I have a TDM400 FXO module connected to a Nokia 32 GSM Terminal ( http://www.nokia.com/nokia/0,,56025,00.html ). Outgoing calls from asterisk to the Nokia work flawlessly. Incoming calls from the Nokia are working fine when asterisk hangs up the call. However, when the far-end hangs up (i.e., the Nokia GSM hangs up the call), asterisk detects the hangup but fails to put the line back onhook and the GSM terminal stays with the line busy until I reload the zaptel/wcfxs modules. I think this is due to some kind of electrical problem and did some voltage measurements by using the debug mode in the wcfxs module. As I understand it, the voltage should be around 48V when the line is onhook. When a call comes in it starts to drop until it reaches 7V, meaning the call is connected. During the call is stays at this value and when the Nokia hangs up it drops the battery for 500ms (this process is called Disconnect Supervision). Asterisk detects this correctly as the Hangup signal and should "open the circuit" so that voltage goes back to 48V indicating the line is onhook and ready for another call. The problem is that the voltage only goes to ~37V instead of 48V and the Nokia terminal still thinks Asterisk has the line offhook. I have included some of these logs. Any help/hints on what causes this problem will be greatly appreciated. I've also looked at the source code of wcfxs.c and despite having a broad idea of how it works, I'm not at all comfortable with messing around with it's low level internals. Hints on any "hacks" to the code that could solve this would be great. Oh, and can anyone tell me what "Debounce" is/does ? Thanks, Leandro -- /var/log/messages -->NOTE: 48V meaning line is onhook and ready for a call Nov 13 19:27:21 raider kernel: Module 2: Installed -- AUTO FXO (FRANCE mode) Nov 13 19:27:21 raider kernel: ProSLIC on module 3, product 0, version 0 Nov 13 19:27:21 raider kernel: Module 3: Not installed Nov 13 19:27:21 raider kernel: Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) Nov 13 19:27:21 raider kernel: Card 3: Voltage: 48 Debounce 0 Nov 13 19:27:21 raider kernel: 7278595 Polarity reversed (0 -> 1) Nov 13 19:27:22 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:24 raider last message repeated 6 times Nov 13 19:27:24 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:27:25 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:25 raider kernel: Setting FXS hook state to 0 (00) Nov 13 19:27:25 raider kernel: Setting FXS hook state to 0 (00) Nov 13 19:27:25 raider kernel: Registered tone zone 2 (France) Nov 13 19:27:25 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:27 raider last message repeated 5 times Nov 13 19:27:28 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:27:28 raider kernel: Card 3: Voltage: 48 Debounce 63 ... Nov 13 19:42:32 raider last message repeated 47 times Nov 13 19:42:33 raider kernel: Card 3: Voltage: 47 Debounce 63 Nov 13 19:42:33 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:42:36 raider last message repeated 6 times -->NOTE: This is when the Nokia rings Asterisk Nov 13 19:42:36 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:36 raider kernel: RING on 2/3! Nov 13 19:42:36 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:37 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:37 raider kernel: Card 3: Voltage: 45 Debounce 63 Nov 13 19:42:37 raider kernel: NO RING on 2/3! Nov 13 19:42:38 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:38 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:38 raider kernel: Card 3: Voltage: 37 Debounce 63 Nov 13 19:42:39 raider kernel: Card 3: Voltage: 28 Debounce 63 -->NOTE: 7V indicates the call is connected Nov 13 19:42:39 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 6 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:41 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:42 raider last message repeated 3 times Nov 13 19:42:42 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:43 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:43 raider kernel: Card 3: Voltage: 6 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:45 raider kernel: Card 3: Voltage: 9 Debounce 63 -->NOTE: At this point Nokia hangs up and drops the battery Nov 13 19:42:45 raider kernel: Battery loss: 2 (63 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 2 (62 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (61 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (60 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (59 debounce) ...
[Asterisk-Users] No audio on outgoing SIP calls over ISDN BRI line
Hello,
I have asterisk installed and working nicely for internal calls using
SIP. However, when I establish an outside call, it rings and connects
properly but I get no audio on either end (the call stays connected).
Asterisk's logs say the following:
-- Executing Wait("Modem[i4l]/ttyI0", "1") in new stack
-- Executing Dial("Modem[i4l]/ttyI0",
"SIP/111|30|Ttr|SIP_CODEC=alaw") in new stack
-- Called 111
-- SIP/111-6e58 is ringing
Mar 30 14:46:27 NOTICE[278545]: channel.c:1451 ast_set_write_format:
Unable to find a path from UNKN to SLINR
-- Got SIP response 603 "Decline" back from 192.168.0.27
== No one is available to answer at this time
-- Executing Hangup("Modem[i4l]/ttyI0", "") in new stack
I suspect it might be a codec related problem (Unable to find a path
from UNKN to SLINR) but as far as I am aware, ISDN uses G.711 and I have
both a-law and u-law activated in asterisk. I wonder why Asterik says
the outside call has an UNKN codec!
The only other possible cause I can think of, is some kind of problem
between my ISDN BRI adapter and Asterisk. I am using isdn4linux. Here
are some logs which might be usefull:
(dmesg)
ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1
loaded
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 1.1.4.1
HiSax: Layer2 Revision 1.1.4.1
HiSax: TeiMgr Revision 1.1.4.1
HiSax: Layer3 Revision 1.1.4.1
HiSax: LinkLayer Revision 1.1.4.1
HiSax: Approval certification failed because of
HiSax: unauthorized source code changes
HiSax: Total 1 card defined
HiSax: Card 1 Protocol EDSS1 Id=HiSax (0)
HiSax: Traverse Tech. NETjet-S driver Rev. 1.1.4.1
PCI: Found IRQ 9 for device 00:0d.0
PCI: Sharing IRQ 9 with 00:04.2
PCI: Sharing IRQ 9 with 00:09.0
PCI: Setting latency timer of device 00:0d.0 to 64
NETjet-S: PCI card configured at 0xb000 IRQ 9
NETjet-S: ISAC version (0): 2086/2186 V1.1
NETjet-S: IRQ 9 count 0
NETjet-S: IRQ 9 count 2
HiSax: DSS1 Rev. 1.1.4.1
HiSax: 2 channels added
HiSax: MAX_WAITING_CALLS added
HiSax: debugging flags card 1 set to 1f
(asterisk)
[chan_modem.so] => (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated
Modem Driver)
-- Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)
== Registered channel type 'Modem' (Generic Voice Modem Channel
Driver)
(lsmod)
Module Size Used byNot tainted
hisax 470288 2
isdn 122688 3 [hisax]
slhc5088 0 [isdn]
(lspci)
00:0d.0 Network controller: Tiger Jet Network Inc. Intel 537
Subsystem: Tiger Jet Network Inc. (Wrong ID) 128k ISDN-S/T
Adapter
Flags: bus master, medium devsel, latency 0, IRQ 9
I/O ports at b000 [size=256]
Memory at df80 (32-bit, non-prefetchable) [size=4K]
Anyone has any clue what might be causing this strange behaviour?
Thanks in advance,
Leandro Morgado
Eurotux / Portugal
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