Re: [asterisk-users] fxotune question
On Wed, Jun 04, 2008 at 11:02:19PM -0400, John Morey wrote: Hello, I've run fxotune Of which zaptel version, exactly? at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. fxotune works by setting the values of some specific registers in a specific chip used for the FXO adapters. In the set mode (-s) it merely takes a set of values from /etc/fxotune.conf and applies them to the chips of the Zaptel device in the respective ports. In the tuning mode (-i or -d) it will attempt to find the best set of register values for your ports. It does that basically by systematically applying many possible sets and and checking the echo level with it. If nothing is connected to the port, no set of value is better than the default values (all zeros), and hence those will remain. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? An echo canceller is still generally useful for an FXO adapter. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] handling SIP trunk with limited concurent calls
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
--
Benoit
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Re: [asterisk-users] Browser based VoIP client?
On Wed, Jun 04, 2008 at 05:48:20PM -0500, Bob G wrote: You can download a FREE browser softphone and or cliick to call that supports UDP athttp://1ezphone.com/download It works well with Asterisk I use it everyday And it's not as if you're affiliated to the company that wrote it, right? I just wonder who was it that posted to the following to the list a few monthes ago: I would like to hire someone to automate my asterisk for hosted PBX service for fetures like user signup, adding money and call bridging Please contact me offline at [EMAIL PROTECTED] In fact, it is not exactly an embedded phone as the OP in the current thread wanted. It is tied up to your specific service. Please keep your messages informative. If you like your software and think it indeed is relevant to subscribers of the list, you can promote it, but keep your messages informative and spam-free. This mailing list is not your marketting page. If you are affiliated with anything, state it clearly. You may use it every day and like it. But you can never be objective about it. Furthermore, your recent messages about the flash phone service have been quite content-free. Writing that it is great just because you say so does not really explain our list readership why they should bother looking at it. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] remote server with Snom 190
I have a local asterisk 1.2 and a remote asterisk 1.4. Snom 190 can be used with the local asterisk but not with the remote one. I need some hints where to track down this issue. Some information: Snom 190: Line 1: Account: 615 Password: OnlyIknowit Registrar: ast.mydomain.com Status: OK Line 2: Account: 6888 Password: Otherside Registrar: 22.33.44.55 (only IP address!) Status: Not found Function keys: P1 Line Number sip:[EMAIL PROTECTED];user=phone P2 Line Number sip:[EMAIL PROTECTED];user=phone Remote server is a fresh installed Ubuntu 8.04 server. What do I miss? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] init.d script no longer uses safe_asterisk
I believe Ubuntu is in the process of migrating from sysvinit to Upstart. Upstart is supposed to be capable of monitoring services to ensure they don't fail, so I suspect this is likely to be the reason behind the safe_asterisk script not being used. Paul Belanger wrote: I noticed safe_asterisk is nolonger used from the init.d script (on ubuntu) for asterisk-1.6.0-beta9. I'm curious if there is another init.d script out there, or even the best way to call safe_asterisk. Or is safe_asterisk nolonger the script of choice for starting, restart asterisk. One of the main reason we like it, is the email notification if it crashes. Thanks, PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:4846b1e767791878237595! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About H323 configuration on Asterix
Hi All, I have an Asterisk IP-PABX which I need to make the H323 channel up with an SBC (ACME). Does anybody have any example configuration guide for this? I am really really new with Asterisk, well PABX in general. So any help will be really appreciated. Thanks in advance. Kr, Sema ARCA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] init.d script no longer uses safe_asterisk
On Thu, Jun 05, 2008 at 03:02:28AM +1000, Rob Hillis wrote: I believe Ubuntu is in the process of migrating from sysvinit to Upstart. Upstart is supposed to be capable of monitoring services to ensure they don't fail, so I suspect this is likely to be the reason behind the safe_asterisk script not being used. upstart can fall back to use init.d scripts. The init.d script of the ubuntu package (and of the Debian one. IIRC it's the same) behaves IMHO better in the presence of safe_asterisk, generates the run dir with proper permissions if it doesn't exist, etc. I'd recommend that you use it rather than the current one from the Asterisk package. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasions where having Oslec and hardware echo cancellation has caused significant problems with audio quality - the usual symptoms are gaps in the conversation as the hardware cancellation eliminates the majority of the echo and the software cancellation subsequently eliminates parts of the conversation. If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? These weren't installs that we'd done - in both instances, they were companies who had bought Trixbox based systems from someone else who had subsequently gone out of business. Granted, I had /assumed/ that OSLEC was causing problems by not disabling itself properly when hardware echo cancellation was found - and I guess that assumption stuck when the removal of OSLEC resolved the problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
On Thu, Jun 05, 2008 at 09:28:52PM +1000, Rob Hillis wrote: Tzafrir Cohen wrote: On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: If you use a hardware EC (or technically: a span-specific echo cancellation method) the generic Zaptel echo canceller (software-based, OSLEC in this case) will not be used. That's not always been my experience with OSLEC. HPEC and the generic Zaptel echo canceller seem to work this way, but as I've said, I've had two cases where I've had to remove OSLEC to stop it degrading voice quality where there is a hardware echo canceller in play. Unless there are some really strange OSLEC patches floating around, of which I'm not aware, this should not be the case. OSLEC uses exactly the same EC interface to Asterisk as the built-in one and HPEC do. Any chance you didn't actually fully unload zaptel? These weren't installs that we'd done - in both instances, they were companies who had bought Trixbox based systems from someone else who had subsequently gone out of business. Granted, I had /assumed/ that OSLEC was causing problems by not disabling itself properly when hardware echo cancellation was found - and I guess that assumption stuck when the removal of OSLEC resolved the problem. Trixbox has (had?) an earlier version of OSLEC that failed to properly handle the echo training function. And it also sets echotraining=800 (which is probably is not such a good idea anyway nowadays). That combination causes bad audio. Disable echo traning and/or upgrade OSLEC. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote server with Snom 190
Ronald Wiplinger wrote: I have a local asterisk 1.2 and a remote asterisk 1.4. Snom 190 can be used with the local asterisk but not with the remote one. I need some hints where to track down this issue. Some information: Snom 190: Line 1: Account: 615 Password: OnlyIknowit Registrar: ast.mydomain.com Status: OK Line 2: Account: 6888 Password: Otherside Registrar: 22.33.44.55 (only IP address!) Status: Not found Function keys: P1 Line Number sip:[EMAIL PROTECTED];user=phone P2 Line Number sip:[EMAIL PROTECTED];user=phone Remote server is a fresh installed Ubuntu 8.04 server. What do I miss? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NAT issues? Is the remote server on a private IP address behind a NAT firewall/router? Firewall issues at either end? At the appropiate ports open on both firewalls for the phone to talk to the remote Asterisk server? Lyle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent, hope your problems go away soon. I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. Currently we have about 200 SIP users which can cause approximately upto 3 simultaneous calls. We are mainly concerned about the performance and stability of asterisk when the load increases. Our server can handle about 100 simultaneous calls having 3Ghz Dual Intl-Xeon Processor with 2GB of ram using G711 codec, and around 30 simultanoeus calls using G729 codec. This we are expecting from the hardware. We are planning to accomodate about 5,000 users on this server. Before the release of 1.6 i heard that its architecture is going to be different from 1.2 and 1.4. Recently i read an article about freeswitchhttp://freeswitch.org/node/117, which explains how its functionality is like asterisk but it can perform better than asterisk due to its architectural differences. The main developer for freeswitch is anthony who also codes for asterisk. He explaines why the architecture of asterisk needs to be changed which requires massive recoding, but nobody took the step to do it. Thats why he started freeswitch on his own to redifine the architecture, so that the performance and reliability of the switch should be better than asterisk. In his article he has already said that freeswitch beats asterisk by a factor of 10. If asterisk architecture is being rewritten in 1.6 to achive the same goal, then we will be happy to use 1.6 instead of shifting the whole system to freeswitch. We dont have any problem or issues with 1.4.2 yet. We are mainly concerned about the its performance when the load increases. If 1.6 is more reliable under heavy loads then we would like to use it. If anyone can put some light on this topic, all i can say is thanx for sharing your thaughts and experiences. On Thu, Jun 5, 2008 at 1:13 AM, Brent Davidson [EMAIL PROTECTED] wrote: Just an update. I tried updating to the newest Rhino Release firmware 1.15 and newest stable driver version 2.2.6. It works OK with zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently running one branch office with the upgraded firmware, driver, zaptel-1.4.9.2 and Asterisk-1.4.20.1. I'll see how everything goes there and may upgrade the other offices if it works OK. Thanks, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default ringtone
Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Hi! I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. [...] We are planning to accomodate about 5,000 users on this server. Many people on this list will advise you to use a SIP proxy like OpenSER in front of Asterisk to take care of SIP registrations - I'll do the same. ;- Before the release of 1.6 i heard that its architecture is going to be different from 1.2 and 1.4. Recently i read an article about freeswitch, which explains how its functionality is like asterisk but it can perform better than asterisk due to its architectural differences. [...] If asterisk architecture is being rewritten in 1.6 to achive the same goal, then we will be happy to use 1.6 instead of shifting the whole system to freeswitch. Looks like you've just decided to move to FreeSwitch ;- While early testing shows that 1.6 can increase SIP performance compared to 1.4 up to factor 3-4, this release is by no means a from-the-ground-up re-write like FreeSwitch. Read more: http://www.voip-info.org/wiki/view/Asterisk+v1.6 http://www.voip-info.org/wiki/view/Asterisk+dimensioning Cheers, Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
Adrian Marsh wrote: Hi All, I’ve trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don’t have any E1/T1 cards at all (am completely IP based). So I don’t think zaptel.conf will come into this (am I right??) I’ve tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I’ve read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I’ve tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
On Thursday 05 June 2008 09:17:49 Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I'd suspect 5-8 have the tip and ring reversed. 9-10 look like they have almost no detected echo at all. I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 You certainly could flip one pair, re-run fxotune (or just manually set it to the same as line 16) and see if the sound dramatically improves on that channel. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
Adrian Marsh wrote: Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. Sorry I don't, wish I could be of more help. I'll see what I can dig up -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
On Thursday 05 June 2008 09:50:05 Eric ManxPower Wieling wrote: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Considering he's using an analog card, that last statement does not apply. Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
So I wonder, is it asterisk itself generating the tones in Dial(), or does it comefom the psedo zaptel driver that generates it ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 16:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. Sorry I don't, wish I could be of more help. I'll see what I can dig up -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] handling SIP trunk with limited concurent calls
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
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[asterisk-users] detecting which party hung up
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten = _9XXX.,n,Dial(${MY_TECH}${MY_NUM}||M(call-answer))
; Trapping call termination here
exten = h,1,NoOp( Call exiting: status ${GLOBAL(${GM})} DS:
${DIALSTATUS} HU: ${HANGUPCAUSE} )
I set the ${GLOBAL(${GM})} variable through a macro 'call-answer', and it
works fine for detecting if the call was answered or not (I have other
logic to run at answer time so it fits me okay).
I thought that there would be a way for me to know on the calling channel
if the 'h' was enetered because this channel hung or because the other
bridged party hung, so I tried ${DIALSTATUS} and ${HANGUPCAUSE}, but they
are always the same no matter who hangs up. Am I missing something here?
Thanks
l.
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] detecting which party hung up
On Thu, Jun 5, 2008 at 6:57 PM, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten = _9XXX.,n,Dial(${MY_TECH}${MY_NUM}||M(call-answer))
; Trapping call termination here
exten = h,1,NoOp( Call exiting: status ${GLOBAL(${GM})} DS:
${DIALSTATUS} HU: ${HANGUPCAUSE} )
I set the ${GLOBAL(${GM})} variable through a macro 'call-answer', and it
works fine for detecting if the call was answered or not (I have other
logic to run at answer time so it fits me okay).
I thought that there would be a way for me to know on the calling channel
if the 'h' was enetered because this channel hung or because the other
bridged party hung, so I tried ${DIALSTATUS} and ${HANGUPCAUSE}, but they
are always the same no matter who hangs up. Am I missing something here?
Thanks
l.
Hi,
add g flag to Dial app, that way Dial will continue to next priority
when ANSWERED but called party hanged up. However if caller will hang
up, channel will jump to h extension.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] handling SIP trunk with limited concurent calls
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit
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[asterisk-users] Asterisk - Nortel CS1K via NRS
Hi, Was wondering if anyone had any tips or experience in getting a Nortel CS1K and Asterisk 1.4.19 to talk to each other via NRS? So far I've gotten asterisk to place calls to the CS1k via the NRS, however calls originated by the CS1K get rejected by the NRS with a 404 Not Found message. If I take the NRS out of the equation by replacing the IP address of the NRS in the CS1K with that of the Asterisk server then everything works ok, however I would like to get the NRS working as it seems to take on the role of SIP proxy server, allowing configuration of multiple SIP trunks where the CS1K seems to be otherwise restricted to a single trunk. Any help appreciated! Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with Polycom phones
I`m curious: did going with numerical IP addresses fix your problem? Mick -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Wednesday, June 04, 2008 13:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Philipp von Klitzing wrote: Hi! I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are using asterisk 1.4.2 for a SIP only based configuration. [...] We are planning to accomodate about 5,000 users on this server. Many people on this list will advise you to use a SIP proxy like OpenSER in front of Asterisk to take care of SIP registrations - I'll do the same. ;- I've seen comments similar to this going around a lot and I've never really understood it. I guess maybe I won't understand it until I am in a situation where I need to handle a huge call volume and hundreds or thousands of users (I.E. probably never). In my situation, using Asterisk as a distributed PBX with Snom SIP phones I haven't had any problems at all with the asterisk end of the system. All of my problems, and I have to stress they are MINOR problems have been related to interfacing to the analog PSTN. I have not not meshed all of the branch offices together yet, so I may run into further issues there, but all of the inter-office calling will be handled by IAX trunking. I really like these Snom 300 phones as far as audio quality goes. I wish they had a few more programmable buttons but that was a purchasing oversight. We underestimated the number of programmable buttons we would need and opted for the 300 instead of the 360. I also wished they used IAX. It's fairly obvious during the software update process thatthey run either a Linux or BSD derivative so it shouldn't be too difficult to develop an IAX firmware for them, even if it has to be done by a third party. I wonder why more vendors haven't adopted IAX yet? -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
I wonder why more vendors haven't adopted IAX yet? I expect that before major players adopt this protocol it'd need to be confirmed as a standard by some form of international body. That was underway, but lacking anyone to push the process along. I would've thought that Digium would be the most likely lead proponent, but that doesn't seem to be the case. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Tilghman Lesher wrote: On Thursday 05 June 2008 09:50:05 Eric ManxPower Wieling wrote: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. Considering he's using an analog card, that last statement does not apply. Drew Gibson wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew Thanks Tilghman, I will try flipping one of the lines at the weekend, can't touch it during the week. Our rxgain was raised from 0.0 to 2.0 at the end of February after complaints that CC staff couldn't hear customers but the 'static' issues pre-date that change. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
On Thu, 2008-06-05 at 13:45 -0500, Michael Graves wrote: I would've thought that Digium would be the most likely lead proponent, but that doesn't seem to be the case. Actually, Digium has been quite active in helping to try to get the IAX protocol adopted as a standard. See http://tools.ietf.org/id/draft-guy-iax-04.txt for the latest draft of the protocol specification as submitted to the IETF. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
5 jun 2008 kl. 20.45 skrev Michael Graves: I wonder why more vendors haven't adopted IAX yet? I expect that before major players adopt this protocol it'd need to be confirmed as a standard by some form of international body. That was underway, but lacking anyone to push the process along. Please note that the IAX draft is just an informational RFC, not anything that goes any IETF standards track or is endorsed by the IETF. There are many vendor-related protocols documented like that. (Said from the chan_sip corner). Cheers, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RECALL: Lithium batteries for Polycom Soundstation 2W
Just released by the CPSC on their recalls mailing list; please forward to any venues where you feel operators or resellers of the SoundStation might be, with this preface included. My 2W had a battery with the part code 1520-07804-002; its date code was GP0806, and therefore predates the recall period. Cheers, -- jra - Begin forwarded message - This message consists of the following: [ 2 unrelated recalls, plus: ] 2. Polycom, Inc. Recalls Wireless Conference Phone Batteries Due to Fire Hazard 2. Polycom, Inc. Recalls Wireless Conference Phone Batteries Due to Fire Hazard NEWS from CPSC U.S. Consumer Product Safety Commission Office of Information and Public Affairs Washington, DC 20207 FOR IMMEDIATE RELEASE June 5, 2008 Release #08-297 Firm's Recall Hotline: (800) 963-7627 CPSC Recall Hotline: (800) 638-2772 CPSC Media Contact: (301) 504-7908 Firm's Media Contact: (925) 924-5689 Polycom, Inc. Recalls Wireless Conference Phone Batteries Due to Fire Hazard WASHINGTON, D.C. - The U.S. Consumer Product Safety Commission, in cooperation with the firm named below, today announced a voluntary recall of the following consumer product. Consumers should stop using recalled products immediately unless otherwise instructed. Name of Product: SoundStation2W Wireless Conference Phones with Lithium Ion Batteries Units: About 5,800 units Phone Distributor: Polycom, Inc., of Pleasanton, Calif. Battery Pack Distributor: Gold Peak Industries Ltd., of Hong Kong Hazard: The battery packs can overheat, posing a fire or burn hazard. Incidents/Injuries: Polycom has received one report of a battery pack overheating resulting in minor property damage. No injuries have been reported. Description: The recalled battery packs were supplied by Gold Peak Industries Ltd. and sold with Polycom's SoundStation2W wireless conference phones from December 1, 2007 until May 2, 2008, and separately as replacement battery packs during the same time period. The SoundStation2W part numbers and SKU numbers are printed on the underside of the telephone and include the following models: Part Number | SKU Number 2201-67800-022 | 2200-07800-001 2201-67880-022 | 2200-07880-001 SoundStation2W recalled battery pack part numbers, SKU numbers and date codes are as follows: Part Number | SKU Number | Date Code 1520-07803-003 | 2200-07803-002 | GP1207, GP0108, GP0208, GP0308 (December 2007 through March 2008) 1520-07804-003 | 2200-07804-002 | GP1207, GP0108, GP0208, GP0308 (December 2007 through March 2008) The battery packs have a black or white plastic coating and a white label with the following title: RECHARGEABLE Li-ion BATTERY. The recalled battery pack part numbers can be found on the bottom right hand corner of the white label on the battery pack. The date code can be found to the left of the part number printed on the white label of the battery. Sold by: Authorized dealers nationwide through catalogs, online, telesales, office supply stores, the Polycom Web store, and Fry's Electronics retail locations from December 2007 through May 2, 2008 for between $700 and $900. Replacement battery packs were sold for between $50 and $90 through the same outlets. Manufactured in: China Remedy: Consumers should immediately remove the battery pack from their SoundStation2W wireless conference phone. Once the battery pack is removed, consumers can still use their conference phone by keeping the charger plugged into the unit. Consumers should not attempt to use battery packs other than those supplied by Polycom in the unit. Consumers should contact Polycom for a free replacement battery pack. Consumer Contact: For additional information, contact Polycom, Inc. at (800) 963-7627 between 9 a.m. and 5 p.m. ET Monday through Friday, or visit the firm's Web site at www.polycom.com/2WBattery To see this recall on CPSC's web site, including pictures of the recalled product, please go to: http://www.cpsc.gov/cpscpub/prerel/prhtml08/08297.html The U.S. Consumer Product Safety Commission is charged with protecting the public from unreasonable risks of serious injury or death from more than 15,000 types of consumer products under the agency's jurisdiction. Deaths, injuries and property damage from consumer product incidents cost the nation more than $800 billion annually. The CPSC is committed to protecting consumers and families from products that pose a fire, electrical, chemical, or mechanical hazard or can injure children. The CPSC's work to ensure the safety of consumer products - such as toys, cribs, power tools, cigarette lighters, and household chemicals - contributed significantly to the 30 percent decline in the rate of deaths and injuries associated with consumer products over the past 30 years. To report a dangerous product or a product-related injury, call CPSC's hotline at
Re: [asterisk-users] Lumenvox - Gentoo
Solved -
I thought I would follow up in case anyone else on the list is using
gentoo. Got some guidance from the gentoo forum. There is a difference in
this function between 1.33 and 1.34 (1.34 is current in gentoo portage)
1.33:
BOOST_FILESYSTEM_DECL bool no_check( const std::string name ); //
always returns true
1.34:
inline bool no_check( const std::string ) { return true; }
As a quick test, I just built 1.33.1 from source and it works. Eventually I
will try emerge =boost-1.33.1 so I can stick with portage. If anyone is on
gentoo currently, be careful about upgrading boost.
Thanks for the reply.
-Kris
On Wed, Jun 4, 2008 at 3:20 PM, David Backeberg [EMAIL PROTECTED]
wrote:
Make sure you enable all the USE flags, and then perhaps try
emerge boost
again
I've had times where leaving out a badly named USE flag meant that
critical things didn't end up getting built. A particularly egregious
must enable all USE flags case is if you try
emerge ejabberd
Without all the USE flags, especially mod_irc (WTF!) you end up with a
useless daemon. Why would you let anybody emerge a chat daemon with no
support for chat?
On Wed, Jun 4, 2008 at 3:34 PM, Kris Edwards [EMAIL PROTECTED]
wrote:
Is anyone running Lumenvox on Gentoo? My asterisk install has been
running
like a champ for a few years now and I really hate the thoughts of
changing
distros just for Lumenvox.
Here is my issue:
The engine needs the libs from boost. I emerged boost and noticed that
there were four libs that the engine were looking for that were not
installed via portage.
libboost_regex.so.2
libboost_thread.so.2
libboost_filesystem.so.2
libboost_date_time.so.2
Instead, I had the above libs without the .2 at the end. I created
symlinks
in the engines lib folder.
Now, when I try to execute the bin I get:
./LVSRE_SERVER: symbol lookup error:
/opt/lumenvox/engine/lib/liblv_lvspeechserver.so: undefined symbol:
_ZN5boost10filesystem8no_checkERKSs
I am using the redhat package. I haven't tried rpath or debian yet
(which
I'm about to do now). Just thought maybe someone might have a thought on
what I should try.
FYI: I also tried un-emerging boost and building directly from the
official
release (1.35 I belive). Perhaps there is a ./configure option I need to
get this to work right. I have little experience with redhat and 0
experience with rpath or debian. I simply used rpm2tar and moved things
appropriately.
Thanks!
-Kris
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--
Kris Edwards
Operations Manager
Impact Radio Group
5660 Franklin Rd. Ste 200
Nampa, ID 83687
(208) 465.9966
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[asterisk-users] Flash Operator panel
Hello My Flash Operator Panel keeps resetting timers everytime i open it or refresh it.. is there a way or config to force it to maintain timers ? _ It’s easy to add contacts from Facebook and other social sites through Windows Live™ Messenger. Learn how. https://www.invite2messenger.net/im/?source=TXT_EML_WLH_LearnHow___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default ringtone
Correct me if I'm wrong, but unless you pass specific options to the dial command to have it override the ringing then when you dial out, you hear the audio from whatever channel you're dialing on. So the tones you are hearing are from the telco. The ring cadences defined in indications.conf are used when Asterisk is ringing a phone connected to an FXS channel. -Brent Adrian Marsh wrote: So I wonder, is it asterisk itself generating the tones in Dial(), or does it comefom the psedo zaptel driver that generates it ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 16:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hmmm.. Well indications.conf does have: country=uk But I've definitly just hearing a long-tone tone, long break, long tone But the file is set to: [uk] description = United Kingdom ringcadence = 400,200,400,2000 ; These are the official tones taken from BT SIN350. The actual tones ; used by BT include some volume differences so sound slightly different ; from Asterisk-generated ones. dial = 350+440 Any idea why? Thanks Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: 05 June 2008 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Default ringtone Adrian Marsh wrote: Hi All, I've trying to force on the ringtone generated for outbound calls with Dial,r but want the tone to be the UK standard. I use Zaptel, but don't have any E1/T1 cards at all (am completely IP based). So I don't think zaptel.conf will come into this (am I right??) I've tried editing zapel.conf anyway, and changed loadzone and defaultzone to =uk I've read through zapara.conf, but cant see a ringtone definition in there. Despite these changes and a restart of zaptel and asterisk via /etc/init.d, I still hear a US ringing sound. So what did I miss? Also, is it possible to generate different ringtones based on dialplan? Eg, if I dial out to a UK number, use the UK ring, but for US use a US one ? In the past I've tried using playtone(), but that stops immediately that the our IP-carrier picks up the call. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users indications.conf is the file you want to edit :) It defines what ringtones and other indication signals to use. Sorry I don't, wish I could be of more help. I'll see what I can dig up ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Similar extension numbers for multiple users
Hi everybody, Is it possible to create similar extension numbers for multiple users. I am looking at a case of virtual PBX with 5 tenants on one server. Any applicable ideas or suggestions would be highly appreciated. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PoE budget
I'm considering using a PoE switch like this... http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=3023334CatId=2800 ...to power as many as 24 Polycom phones of varied kinds. The sales lit indicates 190 watts available for PoE devices. But I'm concerned about a problem someone reported elsewhere... They said... Is there a reason that Polycom phones do not support PoE classes? We ran into a scenario recently where we could only power 11 Polycom 550's on a 24 port switch. This is because the Polycoms do not announce themselves as being in a specific PoE class, even though the phones only need 6W the switch assumes they need as much power as possible and allocates 14.5W to each port. We have had to resort to running unsupported firmware on the switch to get it to power 24 phones. Does anybody here have insight about this? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Similar extension numbers for multiple users
As long as each tenant has its own context you can use the same numbering plan. The only thing you need to keep unique are the names for the SIP devices. If you want your tenants to be able to call each other then you would need to set up a special prefix for each tenant. On Thu, 2008-06-05 at 18:01 -0400, Zeeshan Zakaria wrote: Hi everybody, Is it possible to create similar extension numbers for multiple users. I am looking at a case of virtual PBX with 5 tenants on one server. Any applicable ideas or suggestions would be highly appreciated. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 vs 1.4?
Brent Davidson a écrit : ...I wonder why more vendors haven't adopted IAX yet? Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX as SIP is more mature and reliable in asterisk and zoiper, -- Benoit begin:vcard fn:Benoit Plessis n:Plessis;Benoit email;internet:[EMAIL PROTECTED] tel;home:+33 9 52 49 25 06 tel;cell:+33 6 77 42 78 32 x-mozilla-html:FALSE version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk video alternatives
Hi. At the company I work for, we use Asterisk to communicate with our offices all around the world. Recently, I've been asked to implement a video conference system, asterisk compatible/interoperable as possible. It's preferred but not required to be an open source solution. What options do I have? wich would you suggest me to try? Any good experience with any of these systems? Thanks a lot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] handling SIP trunk with limited concurent calls
Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines if all accounts are 'full'. I've added a call-limit=2 in the
sip.conf entry, but i dont really now how to use it in the dialplan.
ChanIsAvail() was my first try but didn't work.
I've tried chaining Dial() calls:
Dial(SIP/line1/${EXTEN})
Dial(SIP/line2/${EXTEN})
...
but when an error condition occurs (busy/unavailable/whatever) it
dial the same number on every line, which can take a while at the end.
So, is there a way with the DIALSTATUS variable to detect a 'full' peer
?
Yes.
You need to check for CONGESTION.
something like:
n,Dial(SIP/line1/{EXTEN})
n,Noop(Dial line1 failed - we got ${DIALSTATUS})
n,GotoIf($[${DIALSTATUS} = CONGESTION]?tryNext)
n,Hangup
n(tryNext),Dial(SIP/line2/${EXTEN})
But do check that the SIP provider does indeed return CONGESTION ... (You
may not need the call-limit=2, if they check for you, then if at a later
date, they increase the limit, then you don't need to change anything)
Gordon
Isn't there a risk of getting a CONGESTION message from the other party ?
benoit
Another problem i foresee is long delay in dialing sequence when
asterisk will have to dial using 4/5 account
before having a working channel
i think i should look after another sip provider
--
Benoit
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Re: [asterisk-users] fxotune question
The zaptel version is SVN-branch-1.4-r4257 On Thu, Jun 5, 2008 at 2:57 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jun 04, 2008 at 11:02:19PM -0400, John Morey wrote: Hello, I've run fxotune Of which zaptel version, exactly? at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. fxotune works by setting the values of some specific registers in a specific chip used for the FXO adapters. In the set mode (-s) it merely takes a set of values from /etc/fxotune.conf and applies them to the chips of the Zaptel device in the respective ports. In the tuning mode (-i or -d) it will attempt to find the best set of register values for your ports. It does that basically by systematically applying many possible sets and and checking the echo level with it. If nothing is connected to the port, no set of value is better than the default values (all zeros), and hence those will remain. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? An echo canceller is still generally useful for an FXO adapter. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
Drew, I'm also getting complaints of static. Well actually I've complained about it myself and have asked them to have ATT check the lines just to make sure the problem is not on that side. John On Thu, Jun 5, 2008 at 10:17 AM, Drew Gibson [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. Which ports would you expect to be reversed? 5-8 or 9-10? I have similar settings in my fxotune.conf for a TDM2400P and I'm getting complaints of static on the line that I suspect are related to an overtaxed h/w echo canceller My fxotune.conf:- 13=7,255,251,251,2,255,255,1,255 14=9,254,251,255,2,0,1,0,0 15=9,254,251,255,2,0,1,0,0 16=5,0,0,0,0,0,0,0,0 17=9,254,251,255,2,0,1,0,0 18=9,254,251,255,2,0,1,0,0 19=9,254,251,255,2,0,1,0,0 20=9,254,251,255,2,0,1,0,0 21=9,254,251,255,2,0,1,0,0 22=9,254,251,255,2,0,1,0,0 23=9,254,251,255,2,0,1,0,0 24=9,254,251,255,2,0,1,0,0 regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: On June 4, 2008 06:20:57 pm Joe Carroll wrote: Interestingly enough, on the syslog messages from the TNT we are seeing Called = 911, Q850 Cause = 28, SIP Response = 484 That really looks like the switch that the TNT is talking to is rejecting the number, not the TNT... Remember: 9-1-1 is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Similar extension numbers for multiple users
Would it be possible to have a context with includes for each tenant and include that context in the specific tenant contexts that you would have calling each other.if that makes any sense whatsoever.. From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Carlos Chavez [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Similar extension numbers for multiple users As long as each tenant has its own context you can use the same numbering plan. The only thing you need to keep unique are the names for the SIP devices. If you want your tenants to be able to call each other then you would need to set up a special prefix for each tenant. On Thu, 2008-06-05 at 18:01 -0400, Zeeshan Zakaria wrote: Hi everybody, Is it possible to create similar extension numbers for multiple users. I am looking at a case of virtual PBX with 5 tenants on one server. Any applicable ideas or suggestions would be highly appreciated. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnologìa +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client? None of them are very full featured
Wow, rough groupBut good input thanks.I have my tech looking into the user info and CDRs pages.I will keep working on it, thanks agin good input for the most part.I hope some of you downloaded the softphone or clcik to call and tried them.Maybe you could provide with some usefully info, but without stuff like spam emails or dubious claim, but please keep any other stuff you like or dont like coming so I can keep trying. It is for FREE and only for testing. - Original Message - From: Erik Anderson To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Browser based VoIP client? None of them are very full featured Date: Wed, 4 Jun 2008 20:59:28 -0500 On Wed, Jun 4, 2008 at 5:52 PM, Bob G wrote: None of them have features like hold, transfer, voice mail, dtmf, conference as far as I know none of them has caller ID Only 1ezphone.com has all that and the buttons are programmable for CRM features. Hrm: - no apparent compatibility with any service other than that which is offered via 1ezphone - Frequent spammy emails. - on website: ...we are going to make the only phone portal you will every want. - Some poor person's info revealed on the User Account page: http://1ezphone.com/profile.html - Revelation of someone's call history: http://1ezphone.com/callhistory.html# I, for one, won't be giving this a try any time soon. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- See Exclusive Videos: 10th Annual Young Hollywood Awards http://www.hollywoodlife.net/younghollywoodawards2008/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE budget
On Jun 5, 2008, at 5:08 PM, Bill Michaelson wrote: I'm considering using a PoE switch like this... http://www.tigerdirect.com/applications/SearchTools/item- details.asp?EdpNo=3023334CatId=2800 ...to power as many as 24 Polycom phones of varied kinds. The sales lit indicates 190 watts available for PoE devices. But I'm concerned about a problem someone reported elsewhere... They said... -- -- Is there a reason that Polycom phones do not support PoE classes? We ran into a scenario recently where we could only power 11 Polycom 550's on a 24 port switch. This is because the Polycoms do not announce themselves as being in a specific PoE class, even though the phones only need 6W the switch assumes they need as much power as possible and allocates 14.5W to each port. We have had to resort to running unsupported firmware on the switch to get it to power 24 phones. -- -- Does anybody here have insight about this? have used many fsm7326p to power 24 phones or 726tp to power 12 phones and they work great ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk video alternatives
El vie, 06-06-2008 a las 00:24 +0200, Matias Surdi escribió: At the company I work for, we use Asterisk to communicate with our offices all around the world. Recently, I've been asked to implement a video conference system, asterisk compatible/interoperable as possible. It's preferred but not required to be an open source solution. Try vmukti http://sourceforge.net/projects/vmukti/ VMukti is leading Asterisk/ Yate enabled web video conferencing application for Web / PSTN. It is world’s first open source mashable PBX and meeting platform for home and office having features like multipoint audio/ video, desktop sharing, whiteboard. What options do I have? wich would you suggest me to try? Any good experience with any of these systems? I've no tested it before, please let us know your experience using it. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.20.1 with bad gsm file playback
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tilghman Lesher wrote: Well, the issue is that some enterprising person needs to track down exactly which optimization in gcc is causing this problem and point it out to them. We've filed a bug report with them, but without more specific information, their developers aren't going to track it down, either. So it's kind of a stalemate for the time being. Ok, I've got a machine in the lab at the moment that was exhibiting this problem. I'll at least do the -O3 -fno-strict-alising and -O3 -fwrapv changes. Where is the easiest place to set them? export CFLAGS= or will they get overwritten? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFISJHCDQNt8rg0Kp4RAiARAJ0ehcoy6kaAygu8DqM6key53DjxIQCfR0ND ish4puHOI5CIU6gP24F/xHo= =GD0W -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fxotune vs rxgain/txgain
Hi All - I hope somebody can clarify for me what exactly fxotune does, and how it is related to gain settings. I've been reading what appears to be conflicting information from various sources. I've got a box with an AEX800 with 6 lines (from Qwest) running asterisk and zaptel versions 1.4.20.1 and 1.4.11 respectively. We've been experiencing some echo/quality issues on certain calls which seem to happen on all 6 of the lines. I manually calibrated the rxgain/txgain using ztmonitor and a milliwatt test line to the somewhat improbable levels of +10.0/-2.0 (about the same for all 6 lines). These settings yield acceptable call volumes, but echo and noise are problems. If I run fxotune, it gives me the following numbers: 1=10,0,0,0,0,0,0,0,0 2=12,0,0,0,0,0,0,0,0 3=12,0,0,0,0,0,0,0,0 4=10,0,0,0,0,0,0,0,0 5=10,0,0,0,0,0,0,0,0 6=10,0,0,0,0,0,0,0,0 Two questions here: 1) What do these numbers mean? Are they in any way related to either rxgain or txgain? 2) Am I supposed to set rxgain and txgain back to 0 if I use fxotune -s? If I do use these fxotune settings and set rxgain and txgain to zero, the volume on incoming zap calls is almost too low to be heard, but echo issues seem to be solved. Do I have to choose between 1) acceptable call volume with echo or 2) super-quiet call volume without echo? Should I petition Qwest to install a repeater? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
